[kiteki]

Quote:

What he said ^....that's what I meant too: increasing the bitrate of an existing recording does nothing except increase its file size. It won't make it sound any better.
 

Yeah, but increasing the sampling rate does make it sound better.
 

 

[burtomr]

Quote:

Yeah, but increasing the sampling rate does make it sound better.
 

Hmmm...let's be clear -) :
 
Are you sayin' that re-sampling an existing recording to a higher sampling rate will make it sound better?
 
How? It can't gain any more data so the file size will just get bigger and it will sound the same....Right?
 
 
 

 

[elektrikgypsy]

Thanks guys - you've kind of confirmed what I thought. Given that I'm using the E7 d/i into the E9, I can't see that I'd get any appreciable SQ improvement by using the E17 in place of the E7 without going back and re-recording my source material at higher sample rates (with over a year's audio 24/7 that's really not going to happen!).
 
Which begs the question: if there's a market need for this product, and you're all looking forward to it, what are you recording your source material from? Surely not CD at 44.1khz, right?

 

[kiteki]

Quote:

Hmmm...let's be clear -) :
 
Are you sayin' that re-sampling an existing recording to a higher sampling rate will make it sound better?
 
How? It can't gain any more data so the file size will just get bigger and it will sound the same....Right?
 

Just to be clear, there is upconverting, upsampling and oversampling, they are different.
 
If you upconvert a lossy file to lossless, or 128kbps to 320kbps, nothing happens, it's not a .rar file, the info was deleted, I think we are clear there.
 
Oversampling and upsampling adds data which isn't there, yes, this makes the music sound better.
 
For oversampling, try typing "TDA1543" into ebay and you'll find non-oversampling DAC's.
 
For resampling, you can use SoX in Foobar.
 
 

 

[jrkong]

Quote:

Just to be clear, there is upconverting, upsampling and oversampling, they are different.
 
If you upconvert a lossy file to lossless, or 128kbps to 320kbps, nothing happens, it's not a .rar file, the info was deleted, I think we are clear there.
 
Oversampling and upsampling adds data which isn't there, yes, this makes the music sound better.
 
For oversampling, try typing "TDA1543" into ebay and you'll find non-oversampling DAC's.
 
For resampling, you can use SoX in Foobar.
 
 

 
From my understanding, oversampling takes more information from the sample then the DAC needs to function. So by grabbing more information, you get better sound. So from what I understand, you aren't adding data with oversampling but grabbing more information. Upsampling is what i'm most unsure about, but if my logic is right, what upsampling does is "mirror" the frequency to a higher range so that it is easier for the DAC to handle, sounding better as a result.
 
If any of this is wrong please correct me.
 
 

 

[kiteki]

 
I didn't know oversampling takes information that is there, the "anti oversampling" theories that support NOS seem to say advanced algorithms add more information, I'll have to read up on it more I suppose.
 
Upsampling from 16bit/44.1kHz to 24bit/192kHz will add a huge amount of silence between 22kHz and 100kHz, I think this makes it easier for the dac to handle the information or something.
 
It also 'stretches out' the digital waveforms and makes them a bit smoother, need to read more.
 
All I know is from experience both are audible, the HM-601 (using a NOS chip) sounds clearly different than any modern sources I've used.
 
Upsampling using SoX, and also a software driver I have, the difference is very faint, but it is there and sounds pretty cool I think, I like it.
 
Upsampling an actual file though would be pointless, just massive amounts of silence and 10 times larger file size.
 
A real 24/192 recording has information that extends up to 50kHz or higher, not saying anyone can hear it but that's what to look for in the recording to verify if it's legit or upsampled.

 

[jrkong]

Quote:

 
I didn't know oversampling takes information that is there, the "anti oversampling" theories that support NOS seem to say advanced algorithms add more information, I'll have to read up on it more I suppose.
 
Upsampling from 16bit/44.1kHz to 24bit/192kHz will add a huge amount of silence between 22kHz and 100kHz, I think this makes it easier for the dac to handle the information or something.
 
It also 'stretches out' the digital waveforms and makes them a bit smoother, need to read more.
 
All I know is from experience both are audible, the HM-601 (using a NOS chip) sounds clearly different than any modern sources I've used.
 
Upsampling using SoX, and also a software driver I have, the difference is very faint, but it is there and sounds pretty cool I think, I like it.
 
Upsampling an actual file though would be pointless, just massive amounts of silence and 10 times larger file size.
 
A real 24/192 recording has information that extends up to 50kHz or higher, not saying anyone can hear it but that's what to look for in the recording to verify if it's legit or upsampled.

oh i also forgot to ask that oversampling actually reduces the noise produced from a track by distributing it more evenly throughout the track.
 
I found this pic to be a really good representation of how oversampling works:
http://www.audioholics.com/education/audio-formats-technology/upsampling-vs-oversampling-for-digital-audio/image
 

 

[burtomr]

Please correct me if I don't 'get it':
 
Re-sample: Changes stored file size. Typically describes making a large file (flac) into a smaller size (mp3). Re-sampling cannot add data back once it is reduced to a smaller size or format. Re-sampling only makes sense if the goal is to reduce file size and accept the reduction in resolution and sound quality.
 
Over-sample & Up-sample: Describe DAC data processing senarios of existing sound files. Usually done to gain resolution and/or accuracy of the data retrieved from an existing sound file.

 

[jrkong]

Quote:

Please correct me if I don't 'get it':
 
Re-sample: Changes stored file size. Typically describes making a large file (flac) into a smaller size (mp3). Re-sampling cannot add data back once it is reduced to a smaller size or format. Re-sampling only makes sense if the goal is to reduce file size and accept the reduction in resolution and sound quality.
 
Over-sample & Up-sample: Describe DAC data processing senarios of existing sound files. Usually done to gain resolution and/or accuracy of the data retrieved from an existing sound file.

 
You're mixing up "re-sampling" with compression. Re-sampling doesn't exist, you can't re-sample a file. When you compress a file, you take existing information and compress it into a smaller file (FLAC=1141kbps -> mp3 320kbps) so the file goes from 35MB -> 7.48MB.
Sampling has no relative affect on the size of the file to my knowledge...
 
 
When thinking about compression, think of the file or soundwave as a scatter plot graph. When look at the FLAC version or Lossless version of this scatter plot it looks like a complete line, but when you compress it into a smaller file (let's just say to a 320kbs MP3 file) then the scatter plot will only show the most decisive dots.
 
So think of an example where one dot represents 1 KB of information and you are looking 1 second as your range
 
On the FLAC version (if the file has about 5000 kbps) then you will see about 5000 dots in the 1 second range you look at all arranged so the dots are compressed to seem like a line.
When you look at the 320kbs plot, you will see the same trend, but the dots will be more spread out throughout the graph and as a result, you will have a less "informative" graph.
 
Sampling in itself is a whole different matter...
 
But you have the general idea of Oversampling and Upsampling.

 

[burtomr]

Quote:

 
You're mixing up "re-sampling" with compression. Re-sampling doesn't exist, you can't re-sample a file. When you compress a file, you take existing information and compress it into a smaller file (FLAC=8000+kbps -> mp3 320kbps) so the file goes from 35MB -> 7.48MB.
Sampling has no relative affect on the size of the file to my knowledge...
 

Ok, then help me out here:
 
When I use iZotope RX 2 to 'convert' a 24/96 WAV file into a 16/44 WAV, the button is labeled "RESAMPLE". Plus, to quote a LP 'ripper' I know, who does the same thing, says: "iZotope RX Advanced 2.00 used for resampling and dithering"
 
I haven't heard anybody describe 'compressing' a flac to mp3 which might lead to:
 
"...if I only had a bigger 'compressor' I could make the file even smaller" 

?
 
 
 
 
 

 

[jrkong]

Quote:

Ok, then help me out here:
 
When I use iZotope RX 2 to 'convert' a 24/96 WAV file into a 16/44 WAV, the button is labeled "RESAMPLE". Plus, to quote a LP 'ripper' I know, who does the same thing, says: "iZotope RX Advanced 2.00 used for resampling and dithering"
 
I haven't heard anybody describe 'compressing' a flac to mp3 which might lead to:
 
"...if I only had a bigger 'compressor' I could make the file even smaller" 

?
 
 
 
 
 

Ok... most of the terms i'm using comes from a more "musical" perspective. Now I'm not sure if "compression" is a totally proper term to use but MP3 is essentially a compression system for audio, hence my use of the word "compression" another term that would be used in this case would be "bit rate".
 
Now if you were talking about the definition of "resample" and not of the definition that you ended up describing, it would simply mean taking the sample audio track and using it again as an input rather then an output. So in when mixing audio, when one is resampling a track, they would take the master track and reuse it as an input in another track. Though how this is relevant to the iZotope, I'm not entirely sure...

 

[kiteki]

I thought resampling is the process of upsampling or downsampling, that's why it's called SoX resampler - http://sox.sourceforge.net/Main/HomePage
 
Some informative posts on this topic here too - http://www.head-fi.org/t/585763/are-cds-compressed/60#post_7995278
 
 
MP3 is lossy compression, since the data is losst.
 
ZIP is lossless compression.
 
FLAC is lossless compression.
 
WAV is not compressed, unless you go all the way back to the studio, where it was lossy compression so they could fit it on a CD.

 

[jrkong]

Quote:

I thought resampling is the process of upsampling or downsampling, that's why it's called SoX resampler - http://sox.sourceforge.net/Main/HomePage
 
Some informative posts on this topic here too - http://www.head-fi.org/t/585763/are-cds-compressed/60#post_7995278
 
 
MP3 is lossy compression, since the data is losst.
 
ZIP is lossless compression.
 
FLAC is lossless compression.
 
WAV is not compressed, unless you go all the way back to the studio, where it was lossy compression so they could fit it on a CD.

Well i wouldn't be surprised that that was another definition for "resampling." As we all know some words can have more then one definition
 

 

[FoxSpirit]

Quote:

Thanks guys - you've kind of confirmed what I thought. Given that I'm using the E7 d/i into the E9, I can't see that I'd get any appreciable SQ improvement by using the E17 in place of the E7 without going back and re-recording my source material at higher sample rates (with over a year's audio 24/7 that's really not going to happen!).
 
Which begs the question: if there's a market need for this product, and you're all looking forward to it, what are you recording your source material from? Surely not CD at 44.1khz, right?

The E17 supposedly has a much superior DAC section. That it can swallow HD audio is just a side-effect of the better chips. So I'm holding out.

 

[burtomr]

Quote:

Well i wouldn't be surprised that that was another definition for "resampling." As we all know some words can have more then one definition
 

The word 'Resample' is a descriptive term for changing the 'Sample Size' of a digital recording. A typical High Resolution digital recording would be a 192khz/24BPS or 96khz/24BPS. The 'BPS' means Bits Per Sample.
 
When I convert either of those to a 'Redbook' or CD format, which is 44.1khz/16BPS, I 're-sample' the data from 24BPS to 16BPS. This reduces the sample size, the quality, and the resulting file size accordingly.