[adisib]

Can you checksum the before and after files so see if there is any actual change?

 
Did you mis-read what it is supposed to do? The checksum will not change because it isn't changing the file. It basically places the data of the file to contiguous memory on the hard-drive to improve CPU caching. If the operating system starts moving memory around, the memory of the file might get moved around, depending on the operating-system/file-system (NTFS that Windows uses gets defragmented, things like ext4 used by Linux does not get defragmented and much more slowly fragmented), thus "losing the effect". Defragmenting your hard-drive isn't necessarily the same thing I believe, though I'm hardly an expert on file-systems.
 
The same effect could be accomplished just by doing things like loading a file into RAM before playing, which players like Foobar can do. So this is useless to use, even if it did do something, which it doesn't. If you are using a file-system that shouldn't be defragmented, then this wont even do what it is supposed to, and could actually slowly contribute to a more fragmented hard-drive and shorter SSD life.

 

[linglingjr]

The title of this thread "Drastic sound quality improvement" is a complete joke.  
 
From what I understand this program would be useful for loading files onto my ram before being played if I was using a 1 rpm hard drive or several floppy disks, right?

 

[A_Man_Eating_Duck]

Maybe a Mod should close this thread?

 

[WindowsX]

If bits are bits, all machines should sound the same. CD Transport is garbage. Esoteric and all others are selling snake-oil digital CD/computer transports that basically sound the same. Yeah. That sums up of what I read.
 
I wonder how many if you guys are studying about building digital audio transport hardware/software. I'm so fed up with modern machine will handle audio without trouble blablabla for over 10 years. This is sickening.

 

[Fabithierry]

A good digital audio player with support for all kind of audio formats, media library, Wasapi output, SRC and a modern Windows. That's all i need.

 

[WindowsX]

You may want to look into this since it's related to digital transmission
 
http://www.head-fi.org/t/755645/sony-sr-64hxa-low-electronic-noise-microsd-card-for-hq-audio-players#post_11339716
http://www.head-fi.org/t/595071/android-phones-and-usb-dacs/6300#post_11315041
 
But yeah, if you are happy and don't want to try anything new, fine by me. Anything that could passively improving audio without affecting how you use it like this and Fidelizer is recommended.

 

[jiiteepee]

  If bits are bits, all machines should sound the same.

 
Digital machines could sound the same (if bits are not changed) but it's always analog sound you hear so (if you get the bits unchanged to a DAC) only 100% equal DAC conversion quality & anlog stage (everything after the DAC output including the speakers/hedphones) properties can make two system sound the same (if we forget the blacebo effec which may be present even if you think it isn't).

 

[WindowsX]

So all bit-perfect playback sound the same? J River/JPLAY/foobar/etc. using bit-perfect playback should sound the same?

 

[jiiteepee]

  So all bit-perfect playback sound the same? J River/JPLAY/foobar/etc. using bit-perfect playback should sound the same?

 
As digital data does not have sound --> if the bits are not changed by software/hardware and the other components/parameters needed in playback/listening are the same then the playback (sound) is equal. If it doesn't equal then it's not bit-perfect!

 

[WindowsX]

You've made a great misunderstanding statement here. Looks like you don't understand how digital audio works at all. Do you know the concept of timing in digital domain? Have you tried different bit-perfect players like foobar and J River using different bit-perfect audio output like ASIO/WASAPI/Kernel Streaming before? I'm 100% sure that you haven't tried that and I'm also 100% sure that you won't ever try to test them like I said and tell me the result here.

 

[germanium]

  You've made a great misunderstanding statement here. Looks like you don't understand how digital audio works at all. Do you know the concept of timing in digital domain? Have you tried different bit-perfect players like foobar and J River using different bit-perfect audio output like ASIO/WASAPI/Kernel Streaming before? I'm 100% sure that you haven't tried that and I'm also 100% sure that you won't ever try to test them like I said and tell me the result here.

I have done so & have yet to find meaningful differences in spite having a very high resolution system that would put many to shame here. I easily hear differences in different capacitors in the power supply but not bit perfect programs compared to the standard windows player.
 
I'm afraid it is you who is mistaken as there is no timing information other than the order of sample playback until it is streamed from the playback buffer on the sound card. Even then modern DAC's resample the audio data from pulse code modulation to pulse density modulation with sample frequencies up in the megahertz thus making the DAC largely impervious to input jitter. The only jitter that should remain is clock jitter from the local clock on the sound card.
 
Volume control can be easily handled in a perfect manner with 24 bit precision within windows itself so bit-perfect playback is not even all that necessary to get the required sound quality. I even prefer digital volume control to analog volume control for sound quality reasons as it allows the fewest stages of amplification following the DAC. I go directly from my sound card DAC to my amplified speakers with no intervening electronics & found it truly sounds best this way.
 
Differences in audio from digital systems is almost always due to differences in the following analog sections, most other differences are placebo.

 

[WindowsX]

It seems you don't have experience with high quality CD transport/DAC. You should study about time domain in digital audio.

 

[germanium]

  It seems you don't have experience with high quality CD transport/DAC. You should study about time domain in digital audio.

Yes I do have experience in these areas. Enough to know that using separate DAC & transports were problematic especially in the early years. Also Dejittering devices were at times based on a sample rate converter chip that if sample rate conversion was not done then the jitter was in fact worsened not improved, Theta was guilty of this. You see sample rate conversion was necessary with these chips in order to reduce jitter, this was according to the manufacturer of the chip that Theta was using in their dejittering device & since they were not converting sample rate jitter actually went up dramatically, not down. Theta was considered very high quality in their time Other things also appeared to be happening in their dejittering device as well because the sound was completely different before & after. Yes, I listened to it. You were far better off to stick with a qualty box that combined both functions as data transmission protocols were much better than with any protocol used with separate DAC & transports in those days. 

 

[WindowsX]

What about Esoteric/Emm Labs/dCS? Have you tried with one of these gears? Do you understand the concepts of improving clock signal instead of using dejitter methods like masterclock with rubidium/cesium clock? Why would those gears support external clock signal if digital doesn't involve timing?

 

[germanium]

  What about Esoteric/Emm Labs/dCS? Have you tried with one of these gears? Do you understand the concepts of improving clock signal instead of using dejitter methods like masterclock with rubidium/cesium clock? Why would those gears support external clock signal if digital doesn't involve timing?

Yes I have heard of all of those. I have been around digital audio since CD came out in the early 1980's.I do fully understand the clocking concepts The bit about digital audio not involving the clock signal relates to how computers handle digital audio as there in no clock relationship to the final output until the signal is streamed from the soundcard buffer. If you had read the context of what I wrote you would have realized what I was saying. Most separate digital components transferred the clock along with the signal without separation until it reached the DAC's input chip. With this system (SPDIF) there was signal caused jitter. Some higher end brands separated the clock from the signal before leaving the transport & some of these slaved the transport to the DAC clock. Yet others used the I2S signal that is normally used only as a signal protocol within single box players which handled the clock & data signal separately. All of theses were protocols to correct the issues presented by SPDIF signaling. I never said that digital doesn't involve timing just not till the final output from the buffer to the DAC in a computer, this relates only to computer & systems that buffer the data stream.