[Dynobot]

Sounds good, thanks for the info

 

[borrego]

Just back from oversea trip and a busy week, today I spent whole afternoon setting up the DI + NFB1 at home.
 
I encountered many troubles setting up the DI in Windows 7. Using the Windows 7 built-in TE7022 driver result in distortion through WASAPI + Foobar2000 1.1, regardless the buffer size I tried.
 
I ended up using the latest Teradac X2 driver (dated 15th Sep), and setting the Foobar buffer to 150ms to get acceptably smooth sound and consistency (i.e. sound won't distort after playing 3-4 tracks)
 
Another lesson I learnt when trying the Windows 7 audio setting was: Never checking more than the default 48kHz frequency in the "Support Frequency" page. It will cause sound drop off and distortion if 44kHz and 96kHz are also checked.
 
I still think my setup is not optimal as I shouldn't get it to work only with the Teradak driver.
 
Does anyone know if it is normal? Is there a "reference setup" for using the DI with Windows 7 + WASAPI + Foobar2000 1.1?

 

[vrln]

Definately not normal... I had no problems with Win7 64bit, Foobar 1.1 and WASAPI. Anything below 400ms buffer worked fine. Your experiecne sounds very odd actually, maybe there is some kind of a driver conflict? You would try a Linux LiveCD and see if you get any distortion there. Apparently the Tenor chip is well supported in new kernels.

 

[punk_guy182]

Quote:

Is there a "reference setup" for using the DI with Windows 7 + WASAPI + Foobar2000 1.1?

I'd like to know also. I have just installed Windows 7 Professional x64 with foobar2000 v1.1. However, I do not have any sorts of sound drop offs and I have checked both 48kHz and 96kHz as supported formats and I set the default format to 2 channel, 24 bit 96000 kHz.

 

[punk_guy182]

Quote:

Definately not normal... I had no problems with Win7 64bit, Foobar 1.1 and WASAPI. Anything below 400ms buffer worked fine.

At what buffer should I set foobar2000? I noticed that the lower the buffer, the faster foobar responds in WASAPI mode.

 

[ninjikiran]

Thats what a buffer is,
 
Normally USB needs a smaller buffer, why? I don't know why.  All I know is it tends to skip on a higher buffer size.
 
When outputting by spdif, completely different story.  My sound card spdif needs a higher buffer not to skip.
 
Larger buffer size means any audio changes need to be buffered in, like digital volume or extra DSP effects added mid stream.

 

[udailey]

Anyone try DSP1 vs DSP3? I read the description but it leaves me a bit lost as to what they actually DO differently.
Uriah

 

[slim.a]

Quote:

Anyone try DSP1 vs DSP3? I read the description but it leaves me a bit lost as to what they actually DO differently.
Uriah

The DSP1 is a digital filter (oversampling, dithering, filtering...) used with PCM1704 based DACs.

The DSP3 is a DSP used for USB transports (ref3/DI). Its purpose among other things is to reclock the usb stream and allow the possibility for upsampling.
 
The DSP3 and DSP1 cannot be compared directly as they serve different purposes. Having the DSP1 in a DAC doesn't mean you can't benefit or don't need the DSP3 in the DI.
 
In my personal experience, the DSP1's main strength is the quality digital filtering and not its jitter rejection capability. You still need a good transport (whether it is the DI or something else) to get the most from Audio-gd DACs that use the DSP1. The names can be confusing, but there is no direct comparison to be made between DSP1 vs. DSP3.

 

[borrego]

I found the source of my problem this morning. It turns out to be the problem of Foobar2000's replay gain. If I change the replay gain setting, click apply, then there will be distortion and sound drop off. I have to reboot every time if I make change to the replay gain to make it work.
 
The XLR output of NFB-1 is too strong for my Yamaha A-S2000 amp. I have to apply -7db replay gain in Foobar to avoid clipping. CD player connecting to NFB-1 to A-S2000 amp result serious sound clipping. I am emailing Audio-gd for solution.
 
The Yamaha A-S2000's input sensitivity is 2.8V. Seems not that far off from the Krell evolution 302 amp

 

[udailey]

Thanks Slim. That clears it up. 
Uriah
 
Quote:

The DSP1 is a digital filter (oversampling, dithering, filtering...) used with PCM1704 based DACs.

The DSP3 is a DSP used for USB transports (ref3/DI). Its purpose among other things is to reclock the usb stream and allow the possibility for upsampling.
 
The DSP3 and DSP1 cannot be compared directly as they serve different purposes. Having the DSP1 in a DAC doesn't mean you can't benefit or don't need the DSP3 in the DI.
 
In my personal experience, the DSP1's main strength is the quality digital filtering and not its jitter rejection capability. You still need a good transport (whether it is the DI or something else) to get the most from Audio-gd DACs that use the DSP1. The names can be confusing, but there is no direct comparison to be made between DSP1 vs. DSP3.

 

[Marcin_gps]

Hello, 
 
Does the DI support 88.2 and 176.4KHz sample rates?
 
Greets, 
Marcin

 

[ninjikiran]

Negative
 
Quote:

Hello, 
 
Does the DI support 88.2 and 176.4KHz sample rates?
 
Greets, 
Marcin

 

[Pricklely Peete]

 
Quote:

I found the source of my problem this morning. It turns out to be the problem of Foobar2000's replay gain. If I change the replay gain setting, click apply, then there will be distortion and sound drop off. I have to reboot every time if I make change to the replay gain to make it work.
 
The XLR output of NFB-1 is too strong for my Yamaha A-S2000 amp. I have to apply -7db replay gain in Foobar to avoid clipping. CD player connecting to NFB-1 to A-S2000 amp result serious sound clipping. I am emailing Audio-gd for solution.
 
The Yamaha A-S2000's input sensitivity is 2.8V. Seems not that far off from the Krell evolution 302 amp

 
 
AFAIK the balanced inputs should compensate for the higher V of the balanced outs of the source device. Are you sure you have Foobar and Win 7 audio device set up properly ? Get rid of any and all gain leveling, dsp effect, re sampling etc and go with as simple a setup (in the Win7 machine/Foobar) as possible... get yourself a CDP and use it to feed the DAC to trouble shoot your transport ...if the signal sounds normal thorough that (from CDP SPDIF out to the DAC to the Yamaha amp ) then you know something isn't right with the computer setup.
 
Peete.

 

[borrego]

 
Quote:

 
 
 
AFAIK the balanced inputs should compensate for the higher V of the balanced outs of the source device. Are you sure you have Foobar and Win 7 audio device set up properly ? Get rid of any and all gain leveling, dsp effect, re sampling etc and go with as simple a setup (in the Win7 machine/Foobar) as possible... get yourself a CDP and use it to feed the DAC to trouble shoot your transport ...if the signal sounds normal thorough that (from CDP SPDIF out to the DAC to the Yamaha amp ) then you know something isn't right with the computer setup.
 
Peete.

No replay gain, playback buffer and dsp WAS my previous Foobar setup. Unfortunately now I have to apply replay gain but it seems the DI does not like it too much.
 
I already mentioned I tried the CD player through coaxle and optical and there were clippings.
 
Kingwa replied my email and said the NFB-1 output 5V max and my A-S2000 takes 2.8V max. Kingwa suggested to trim the XLR signal to about 2V. Now I am considering buying the TC Electronics Level Pilot. But it may cause effect to the sound.

 

[Caphead78]

"CD player connecting to NFB-1 to A-S2000 amp result serious sound clipping. I am emailing Audio-gd for solution."
 
I think he has tried the CD player idea already
 
EDIT: Whoops looks like I was a little late
Quote:

get yourself a CDP and use it to feed the DAC to trouble shoot your transport ...if the signal sounds normal thorough that (from CDP SPDIF out to the DAC to the Yamaha amp ) then you know something isn't right with the computer setup.
 
Peete.