[Olias of Sunhillow]

Quote:

Exactly! People are wanting the NFB-12 to sound like something its not and wasn't designed to sound like. The NFB-12 is smooth, period. That's why I have said all along that it would be a good match for Beyers and Grados.

Agreed... this is exactly why I chose the NFB-12 to match with my RS-1s at work.
 
Now, in practice, the NFB-12 turns the RS-1s into something I never thought they'd be: smooth and balanced cans that I can listen to for hours at a time. I wouldn't want the NFB-12 to be my only Grado amp -- they don't rock quite hard enough for those moments when you really want to crank it -- but for my office rig and all-day listening, the NFB-12 is perfect so far.

 

[dw1narso]

 
Quote:

Quote:

Exactly! People are wanting the NFB-12 to sound like something its not and wasn't designed to sound like. The NFB-12 is smooth, period. That's why I have said all along that it would be a good match for Beyers and Grados.

Agreed... this is exactly why I chose the NFB-12 to match with my RS-1s at work.
 
Now, in practice, the NFB-12 turns the RS-1s into something I never thought they'd be: smooth and balanced cans that I can listen to for hours at a time. I wouldn't want the NFB-12 to be my only Grado amp -- they don't rock quite hard enough for those moments when you really want to crank it -- but for my office rig and all-day listening, the NFB-12 is perfect so far.

Olias, what do you think of NFB-12 paired with HF2?
 
HF2 is not usual Grado and is smooth already naturally... would NFB-12 still have the grunt to make it alive?

 

[Olias of Sunhillow]

Quote:

Olias, what do you think of NFB-12 paired with HF2?
 
HF2 is not usual Grado and is smooth already naturally... would NFB-12 still have the grunt to make it alive?

Hello! Good question. I haven't tried this combo yet, as my HF-2s stay home and the NFB-12 stays at work. I'll take the HF-2s into the office for a few days and get back to you....

 

[dw1narso]

 
Quote:

Hello! Good question. I haven't tried this combo yet, as my HF-2s stay home and the NFB-12 stays at work. I'll take the HF-2s into the office for a few days and get back to you....

Thanks... I need to know that... cause that's what I plan to pair the NFB12 with...
 

 

[supercurio]

Quote:

udial at 44k, sampled at 96k :
udial upsampled at 96k, sampled at 96k:
 
will like to ask, how do u go about setting upsampled, and the sampled frequency? will like to find out the method u used, jus to rule out software resampling which took place in the OS audio path.

 
Hi!
 
In the previous graphes:

1. 44k were untouched samples until arriving to the DAC.
2. 44k upsampled were upsampled to 96k with Foobar + SoX plugin very high quality (sox resampling is the best rate conversion I know).

 
 
Both were sampled at 96k by an Asus Xonar Essence ST

 

[PPkiller]

scenario A
44K untouched --> Asus Xonar upsampled to 96k --> nfb-12
 
Scenario B
44K upsampled to 96k by foobar + SoX --> Asus Xonar(since already 96k so i guess no resampling) --> nfb-12
 
 
i will guess scenario A, Asus xonar is doing the resampling which could contribute to the sampling graph that u have generated.
 
Are you able to do the following?
 
44K untouched --> Asus Xonar configured for 44k output --> NFB-12
 
with foobar + wasapi
 
Quote:

Quote:

udial at 44k, sampled at 96k :
udial upsampled at 96k, sampled at 96k:
 
will like to ask, how do u go about setting upsampled, and the sampled frequency? will like to find out the method u used, jus to rule out software resampling which took place in the OS audio path.

 
Hi!
 
In the previous graphes:

1. 44k were untouched samples until arriving to the DAC.
2. 44k upsampled were upsampled to 96k with Foobar + SoX plugin very high quality (sox resampling is the best rate conversion I know).

 
 

Both were sampled at 96k by an Asus Xonar Essence ST

 

[supercurio]

Quote:

scenario A
44K untouched --> Asus Xonar upsampled to 96k --> nfb-12
 
Scenario B
44K upsampled to 96k by foobar + SoX --> Asus Xonar(since already 96k so i guess no resampling) --> nfb-12
 
 
i will guess scenario A, Asus xonar is doing the resampling which could contribute to the sampling graph that u have generated.
 
Are you able to do the following?
 
44K untouched --> Asus Xonar configured for 44k output --> NFB-12
 
with foobar + wasapi
 
Quote:

Quote:

udial at 44k, sampled at 96k :
udial upsampled at 96k, sampled at 96k:
 
will like to ask, how do u go about setting upsampled, and the sampled frequency? will like to find out the method u used, jus to rule out software resampling which took place in the OS audio path.

 
Hi!
 
In the previous graphes:

1. 44k were untouched samples until arriving to the DAC.
2. 44k upsampled were upsampled to 96k with Foobar + SoX plugin very high quality (sox resampling is the best rate conversion I know).

 
 

Both were sampled at 96k by an Asus Xonar Essence ST

 

 
Much simpler that that, as described: sorry I don't understand your virtual setup
 
Once again:
 
44k: untouched, which means
44k file => 44k SPDIF => 44k NFB-12 DAC => 96k ADC to do the graph (Xonar Essence ST)
 
44k 96k upsampled
44k file => upsampled by sox to 96k => 96k SPDIF => 96k NFB-12 DAC => 96k ADC to do the graph (Xonar Essence ST)
 
(yes, artifacts at 44k are due to NFB-12)
PS: I used 2 computers, one to play / drive the SPDIF , one other to record.

 

[PPkiller]

are u able to generate a 48k graph using the same setup?
 
Quote:

Quote:

scenario A
44K untouched --> Asus Xonar upsampled to 96k --> nfb-12
 
Scenario B
44K upsampled to 96k by foobar + SoX --> Asus Xonar(since already 96k so i guess no resampling) --> nfb-12
 
 
i will guess scenario A, Asus xonar is doing the resampling which could contribute to the sampling graph that u have generated.
 
Are you able to do the following?
 
44K untouched --> Asus Xonar configured for 44k output --> NFB-12
 
with foobar + wasapi
 
Quote:

Quote:

udial at 44k, sampled at 96k :
udial upsampled at 96k, sampled at 96k:
 
will like to ask, how do u go about setting upsampled, and the sampled frequency? will like to find out the method u used, jus to rule out software resampling which took place in the OS audio path.

 
Hi!
 
In the previous graphes:

1. 44k were untouched samples until arriving to the DAC.
2. 44k upsampled were upsampled to 96k with Foobar + SoX plugin very high quality (sox resampling is the best rate conversion I know).

 
 

Both were sampled at 96k by an Asus Xonar Essence ST

 

 
Much simpler that that, as described: sorry I don't understand your virtual setup
 
Once again:
 
44k: untouched, which means
44k file => 44k SPDIF => 44k NFB-12 DAC => 96k ADC to do the graph (Xonar Essence ST)
 
44k 96k upsampled
44k file => upsampled by sox to 96k => 96k SPDIF => 96k NFB-12 DAC => 96k ADC to do the graph (Xonar Essence ST)
 
(yes, artifacts at 44k are due to NFB-12)
PS: I used 2 computers, one to play / drive the SPDIF , one other to record.

 

[thuantran]

@supercurio: which output method are you using with foobar? If it is DS you can't be sure that the sampling rate reaching the NFB-12 is what you set in foobar player. Say you have 44.1KHz from foobar goes to Xonar set at 44.1KHz but inside Windows Playback Devices Control panel you set your Xonar output to something different then Windows will resample the output. Unless you're using WASAPI you have to set sampling rate at 3 places (foobar, Xonar, Windows playback devices control panel) to be the same.

 

[madwolf]

Quote:

update, the center contact on the switch is ground.  technically one could lift both pin 22s, then connect to them together and run a single 30awg wire over to the bottom contact on the input select switch.  this would give you the correct filter for 44.1 usb and 96k for opt/coax.
 
i may have some ceramic pad isolators around here somewhere to properly isolate and locate pin 22 safely.
 
just general note... this thing is pretty well put together for it's price range.  I am J-Standard Class III Cert. and it is definitely above Class I work, it is Class II+ looking at first glance (no scope and no extra lighting used at the moment).  It has good part placement, little light on the solder in some places, some left over flux in some spots, and 1 solder splat was found stuck to a capacitor.  
 
Now here's my comment.  I want to see this run on a device that has the BW to actually RMAA this thing.  Good efforts so far to show the filter options work, but we don't want people taking those graphs to heart.

The NFB12 comes with a number of power regulator for the different section, if you want to tap the ground/Vcc it would be advisable to choose the Digital ground used with the 8741, I choose a very close capacitor which is connected to both pin 7 and 8 of the 8741. 
 
I am not even J-standard (any class) certified   

  
 
If you would sponsor me a device with a BW to RMAA this I would gladly do so again 

 
 
 
 


Quote:

The high frequency roll off define sound flavor, it bring a smooth flavor.
But the roll off not mean its sound unneutral, the neutral is depend on the circuits components less coloration, like some tube gears even without roll off at 20KHz but sound tube like (coloration).
 

 
You given us a bonus actually, or rather two bonus
There are 3 sound signature for the NFB12 
Is anyone complain about receiving 2 extra  

 
The 3 sound signature could be set by using different input data rate. 
Maybe using 44 is the "smooth musical" sound flavor
using 96 is in between "musical and neutral "
and 192 is "neutral "

 
Quote:

@supercurio: which output method are you using with foobar? If it is DS you can't be sure that the sampling rate reaching the NFB-12 is what you set in foobar player. Say you have 44.1KHz from foobar goes to Xonar set at 44.1KHz but inside Windows Playback Devices Control panel you set your Xonar output to something different then Windows will resample the output. Unless you're using WASAPI you have to set sampling rate at 3 places (foobar, Xonar, Windows playback devices control panel) to be the same.

 
As long as the output is 192 it is fine. 
 
Maybe supercurio could use his Xonar (since my X-fi is substandard) to record a part of a song in a set up like 
PC (playback) -->NGB12-->Xonar-->PC (to record)
You can then record the 3 different sound flavor and share so that other could hear the difference. 
 
 

 

[LiqTenExp]

Quote:

Quote:

update, the center contact on the switch is ground.  technically one could lift both pin 22s, then connect to them together and run a single 30awg wire over to the bottom contact on the input select switch.  this would give you the correct filter for 44.1 usb and 96k for opt/coax.
 
i may have some ceramic pad isolators around here somewhere to properly isolate and locate pin 22 safely.
 
just general note... this thing is pretty well put together for it's price range.  I am J-Standard Class III Cert. and it is definitely above Class I work, it is Class II+ looking at first glance (no scope and no extra lighting used at the moment).  It has good part placement, little light on the solder in some places, some left over flux in some spots, and 1 solder splat was found stuck to a capacitor.  
 
Now here's my comment.  I want to see this run on a device that has the BW to actually RMAA this thing.  Good efforts so far to show the filter options work, but we don't want people taking those graphs to heart.

The NFB12 comes with a number of power regulator for the different section, if you want to tap the ground/Vcc it would be advisable to choose the Digital ground used with the 8741, I choose a very close capacitor which is connected to both pin 7 and 8 of the 8741. 
 
I am not even J-standard (any class) certified   

  
 
If you would sponsor me a device with a BW to RMAA this I would gladly do so again 

 
 
 
 


Quote:

The high frequency roll off define sound flavor, it bring a smooth flavor.
But the roll off not mean its sound unneutral, the neutral is depend on the circuits components less coloration, like some tube gears even without roll off at 20KHz but sound tube like (coloration).
 

 
You given us a bonus actually, or rather two bonus
There are 3 sound signature for the NFB12 
Is anyone complain about receiving 2 extra  

 
The 3 sound signature could be set by using different input data rate. 
Maybe using 44 is the "smooth musical" sound flavor
using 96 is in between "musical and neutral "
and 192 is "neutral "

 
Quote:

@supercurio: which output method are you using with foobar? If it is DS you can't be sure that the sampling rate reaching the NFB-12 is what you set in foobar player. Say you have 44.1KHz from foobar goes to Xonar set at 44.1KHz but inside Windows Playback Devices Control panel you set your Xonar output to something different then Windows will resample the output. Unless you're using WASAPI you have to set sampling rate at 3 places (foobar, Xonar, Windows playback devices control panel) to be the same.

 
As long as the output is 192 it is fine. 
 
Maybe supercurio could use his Xonar (since my X-fi is substandard) to record a part of a song in a set up like 
PC (playback) -->NGB12-->Xonar-->PC (to record)
You can then record the 3 different sound flavor and share so that other could hear the difference. 
 
 

If one were simply looking to ground the pin yes, but if you looked I was not.
 
J Standard is a soldering standard BTW 

 

 

[Kingwa]

If you are knowning the digital filter technology and practise you will don't surprise a lot digital audio gears had different sonice at different oversample rates. I am not a teacher and my English is not allow I explain the detail. I will try to find time interpret my Chinese article maybe can explain clearer.
 Search diagram in the link maybe help.
http://www.stereophile.com/digitalprocessors/

 

[Audiophil]

Hi Kingwa,
nice to have you in the discussion.
I have been looking to > 10 devices's measurements from your link. Naturally most of them have different curves on the different sample rates. But also nearly all of them has a roll-off on 44.1 KHz of max. -1 dB @ 20kHz. This means no difference in sound footprint.
This is different in NFB-12. I assume (as I can only assume) the sonic character as you are describing it, is related to fs=44,1 kHz. I believe at higher rates it will sound different, right?

 

[Kingwa]

The R2R chips usually had close sonic at different oversampling rate.
But the lot sigma-delta chips is different sonic at different oversampling rate because the  different oversampling digital filter applied.
 
May I put the WM8741 as a example.
Case A, the WM8741 fix to a single type digital filter, it had different roll off at high frequency cause the sound flavors had some different.
Case B, The WM8741 applied different type digital filter, it had less different  roll off at high frequency, but it applied the 8X oversampling at 44/48K, and 4X oversampling at 88/96K, and 2X oversampling at 176/192K, also cause the sound flavors had some different.
 
Except the different oversampling filter applied cause different sound, in different oversample rates input, the DAC must had different frequency band also can cause the sonic different.(except the NOS)
 
Then we want to ask, which case cause the more sound different between the different roll off at high frequency  and the 2X/4X/8X oversampling digital filter applied?
 
I think the different oversampling filter sound is quite different. Lower oversampling sound usually  smoother than higher oversampling . So some people prefer NOS butNOS had quite botchy  specs .
 
For the sonic listen , we choice the analog filter caps large than the NFB11, just for smooth sound and reduce the sound different between different oversampe rate input.
 
 
Do you understand my broke English explain? I want to explain more but I can't find the words for use.

 

[Audiophil]

yes, I think I understand in general. For sure some of the experts in the forum have expertise how to deal with the specific technique, me not really.
But no doubt, different filters and oversampling rates have influences and I'm sure you checked carefully.
I was simply surprised, when I recognized the drop-off @ 44,1 kH and was not sure if the design would be nicely balanced.