96 khz sample rate wav files

[QUOTE=tubebar;2556798]You can’t, CD only supports 16 (not anything else). If you are asking about upsampling to 24bit it’s a waste of space as the file is already 16bit…I am sure foobar 2k will do it, give it a try the sound won’t change.

If you want to make a playable disc w/ 24bit you’ll need to use DTS or DVDA which your player has to support, you can’t do CDDA. As I said upsampling is a waste of space.[/QUOTE]

Thanks for the info tubebar :slight_smile:
and I got this


it says
[B]The RCD-W222ES does an exceptional job recording from analog sources. The 24-bit A/D converter and Super Bit Mappingâ„¢ technology ensures a smooth transition from analog to digital. Super Bit Mapping uses noise-shaping technology to approximate 20-bit performance from a regular 16-bit CD. You’ll enjoy more detailed sound and increased dynamic range![/B]
I try to have some burn after thanksgiving

[QUOTE=Burnsama;2556780]How to create 16bit CD to 24bit CD ?
TIA :)[/QUOTE]
As stated above, there really is no reason to store more bits than the original, if your final goal is standard audio CD…Stick to 16bit/44.1 khz

[B]“20-bit performance from a regular 16-bit CD”.[/B]

Nonsense!..Compare an original 24bit and a 16bit of the same recording…Do it blindly(ABX) test…Try with f2k’s utility…
http://www.foobar2000.org/components/view/foo_abx

[QUOTE=t0nee1;2556850]As stated above, there really is no reason to store more bits than the original, if your final goal is standard audio CD…Stick to 16bit/44.1 khz

“20-bit performance from a regular 16-bit CD”.
Nonsense!..Compare an original 24bit and a 16bit of the same recording…Do it blindly(ABX) test…

http://en.wikipedia.org/wiki/Red_Book_(audio_Compact_Disc_standard)[/QUOTE]

thanks for the link t0nee1 :slight_smile:
need time for study :o

Yep that’s a silly gimic, do you understand how audio sampling works?

When you digitally record from an analog source (truly lossless) at 44.1 / 16 that means it is “sampling” at 44,1000 times per second (think of it recording ON and OFF extremly FAST). Then it selects samples to match, in this case it’s 16bit so it chooses from 16000 diff samples.

Upsampling can’t add samples or record something that isn’t there! It’s like converting MP3 lossy back into a WAV the info simply isn’t there - IMPOSSIBLE!

Yes that is BS, sure you can use tricks to make it sound different (louder, EQ etc.) but you cannot add data (sounds that were never recorded in the first place).

16bit so it chooses from 16000 diff samples.

Actually 65536 (2^16) sampling levels.

[QUOTE=olyteddy;2557016]Actually 65536 (2^16) sampling levels.[/QUOTE]

I didn’t know the exact ammount so thanks for the info. :bigsmile:

More info here

You notice as they say “24bit Recording” not upsampling. Yes if you record from the source at 24bit it certainly improves the quality. If that source is 16bit it won’t do anything.

Interesting article thanks.

[QUOTE=tubebar;2557063]if you record from the source at 24bit it certainly improves the quality. If that source is 16bit it won’t do anything.[/QUOTE] except increase storage space in ever so wasteful ways.

If you are resampling to a different frequency then you are losing sound accuracy (because you are interpolating, ie making estimates/weighted averages of multiple individual samples), if you are converting to a different bit depth, you are either making no change to the accuracy (if increasing bit depth) or losing accuracy (if decreasing bit depth).

Once you’ve sampled a source audio waveform at a particular sampling frequency and bit depth, nothing you can do to the digital recording will make it any more accurate than the original recording.

There are tricks like multi-sampling (at sound reproduction stage), noise removal and other tricks you can do to make it appear to sound better, but it can never actually be better than the original first recording (at the original specific sampling frequency & bit depth).

The only time it makes sense to change the frequency/bit depth of a recording is when you are mixing multiple sources.

Thanks for the info tubebar and debro :wink:
any idea make better sound my original CD collection ?

[QUOTE=debro;2557080]
If you are resampling to a different frequency then you are losing sound accuracy (because you are interpolating, ie making estimates/weighted averages of multiple individual samples), if you are converting to a different bit depth, you are either making no change to the accuracy (if increasing bit depth) or losing accuracy (if decreasing bit depth). [/QUOTE]

It shouldn’t change at all since it’s just PCM. From all my tests all it did was add 0000’s in the extra space, the freq. analysis has always been unchanged. If I took a screen shot they’d look identical (in Nuendo or Audtion).

Upsampling PCM should not actually take away anything (won’t add anything either). Downsampling and dithering etc. is another story.

[QUOTE=Burnsama;2557136]Thanks for the info tubebar and debro :wink:
any idea make better sound my original CD collection ?[/QUOTE]

Depends what you mean by better? I guess you can try the fake MP3 SFX which converts ST to Surround (fake) you need a player that supports it. http://www.all4mp3.com/Learn_mp3_sx_1.aspx

As was said NO there isn’t anyway to make it sound better, different there’s many ways.

I don’t have any MP3 file :slight_smile:

[QUOTE=tubebar;2557230]It shouldn’t change at all since it’s just PCM. From all my tests all it did was add 0000’s in the extra space, the freq. analysis has always been unchanged. If I took a screen shot they’d look identical (in Nuendo or Audtion).

Upsampling PCM should not actually take away anything (won’t add anything either). Downsampling and dithering etc. is another story.
[/quote]
Yes.
However, when changing the sampling frequency (either increasing or lowering), the program is estimating the value between two samples using an algorithm which may, or may not, be greatly improved for accuracy within a few years.

Either way, it’s no improvement to the original source now, but in a few years, may be significantly worse :wink:

[QUOTE=tubebar;2557230]Depends what you mean by better? I guess you can try the fake MP3 SFX which converts ST to Surround (fake) you need a player that supports it. http://www.all4mp3.com/Learn_mp3_sx_1.aspx

As was said NO there isn’t anyway to make it sound better, different there’s many ways.[/QUOTE]

There are a few ways to make audio cd’s sound better.

  1. Better quality speakers.
  2. Better quality (lower noise) sound card/player/amp.
  3. Mute all non-essential sound sources (line in/microphone/computer sounds/etc) leaving just the CD input being mixed.
  4. Change your sound card drivers - sometimes drivers are just plain bad (I’m looking at you Creative - Sound Blaster).

[QUOTE=debro;2557263]Yes.
However, when changing the sampling frequency (either increasing or lowering), the program is estimating the value between two samples using an algorithm which may, or may not, be greatly improved for accuracy within a few years.

Either way, it’s no improvement to the original source now, but in a few years, may be significantly worse :wink:
[/QUOTE]

Downsampling a freq is going to lose quality (and lose the freq), you always lose when you go down. Again it shouldn’t make any diff in going up. Yes if you go back down you will lose quality compared to the original won’t be 1:1 if that’s what you’re saying?

You completely lost me when talking about years? How would time change a data file? Aside from errors that can be avoided w/ proper archiving.

[QUOTE=tubebar;2557266]Downsampling a freq is going to lose quality (and lose the freq), you always lose when you go down.
[/quote]
When changing to lower frequency, you are losing your higher frequencies, and then estimating the remaining frequencies.

[QUOTE=tubebar;2557266]Again it shouldn’t make any diff in going up. Yes if you go back down you will lose quality compared to the original won’t be 1:1 if that’s what you’re saying?
[/quote]
You will always lose quality, because you will always be [U][I][B]estimating[/B][/I][/U] the value of the new sample depending on the time when the [I]new [/I]sample falls between the two [I][B]REAL [/B][/I]samples, and the value that the new sample takes will be a sample of a RECONSTRUCTED waveform.

The new sample will be from a reconstructed (ie; estimated) waveform, and is, therefore, not accurate.

While waveform reconstruction algorithmns these days are pretty good, and you are unlikely to hear the difference, you have still lost accuracy.

In 5 years time, the reconstruction algorithmns may be using increased higher orders, which provides better (more accurate) reconstruction … however, you’ve already resampled your original waveform with an older algorithmn with a lower order, and probably deleted the original as well … therefore, you’re forever stuck with an estimate of the original recorded waveform.

In the event that you resample to a multiple frequency (lets assume 2x), the algorithm spits out a new sample between the two REAL samples, which is the closest representable sample of the reconstructed waveform between the two REAL samples [I]based on the algorithms used today[/I].

When you reconstruct the waveform (for playback), the player is interpolating (ie; estimating) values between a REAL value, and an interpolated value … which is less accurate then interpolating between two real values.

Again, if reconstruction algorithms are improved between now and a few years in the future and you’ve resampled your original recording to a higher frequency sample rate … you’re stuck with the crappy estimate.

10 years ago, most wave reconstruction was basically linear on cd players, and premium players reconstructed waveform based on squared reconstruction (2nd power), whereas now, it’s based on a cubed reconstruction (3rd power)… and on that it could be even higher power filters. I’m not entirely sure, because I haven’t been following CD players recently, and have neglected my studies on where audio processing software has gone in the last 5 years :wink:

As technology progresses, processing power increases and hence sound reproduction quality increases …

[QUOTE=tubebar;2557266]You completely lost me when talking about years? How would time change a data file? Aside from errors that can be avoided w/ proper archiving.[/QUOTE]
Resampling to different frequencies = loss of quality, and possibly accidental removal of higher (audible) frequencies.
Resampling to lower bit depth = HUGE loss of quality.
This discussion excludes the accuracy of storage devices.

Thanks for the explanation debro, what you are saying makes sense now ;). I see where you’re getting at. I didn’t quite realize that upsampling would effect the quality, but I always thought it would. I knew this was the case w/ a lossy format (mp3 to wav) but didn’t see how (or see in my tests) on upsampling but it makes sense.