Why would a PC be better?

avahifi · « **Reply #40 on:** 29 Apr 2005, 06:15 pm »

DKW admits I am not wrong after all, its just a matter of how many bits you throw away with what quality equipment before there is any "significant impact".

Oh well, I still wonder how this compares to the "significant impact" of changing brands of computer power cords.

By the way, what clock speed do you need to recover all the bits (2 to the 16th power) at high frequencies with a one bit (serial) processor on a "one pass" basis without sample and hold or noise shaping circuits? Are there any one bit playback units with the necessary clock speed?

Frank Van Alstine

srclose · « **Reply #41 on:** 29 Apr 2005, 06:42 pm »

Quote from: goskers

If you could try to do some strict listening tests to determine if the pc as transport is better with or without the off ramp then that would be great. I have not used the SB2 yet but have had some email conversations with the slim devices guys. They are listening to their customers quite a bit about product development and that seems very evident with the new offering.

I'll be back with some details Sunday.

srclose · « **Reply #42 on:** 1 May 2005, 09:17 pm »

Just left a description of the Off-Ramp in the Empirical Audio circle.

http://www.audiocircle.com/index.php?topic=18767.msg164959#164959

jhenderson010759 · « **Reply #43 on:** 2 May 2005, 01:07 am »

Quote from: goskers

Anybody had any experience with the best digital out from a pc as their goal?

I have used quite a few variations.

Currently, I am using an SB1 which is my preferred solution due to convenience of SlimServer in conjunction with TelCanto for PDA. I have an SB2 on order. But the only reason I am upgrading to an SB2 is because it performs digital attenuation (volume control) using 24-bit operations, as opposed to the SB1's 20-bit. This yields 48 rather than 24 bits of useful remote-control volume via SlimServer and digital out without any signal integrity degradation.

I also have an EMU 0404. And a Roland M1000 digital mixer which features USB soundcard drivers. And a Yamaha DP-U50 USB soundcard. All of which I have used via Foobar, in 24-bit mode.

All of these solutions sound identical when used in an all-digital signal path (up to the amps), as I do. As well they should, since these are all bit-perfect devices (as verified by me in RMAA testing).

Each of these devices has sub 20 nS jitter specifications - some far lower. A large, conclusive body of technical literature has proved beyond any shadow of a doubt that jitter levels at or below this threshold are inaudible. However, I realize that most audiophiles could care less about the audibilty - it's all about vanishingly lower jitter specifications.

Each of these solutions are measureably and audibly equivalent. Pick one or any other which best suits your convenience requirements.

WerTicus · « **Reply #44 on:** 3 May 2005, 05:54 pm »

i just ordered a lynx l22

ill lets you all know how she is.

vs an 18k bel canto player

Carlman · « **Reply #45 on:** 3 May 2005, 06:17 pm »

My guess is that the Bel Canto will win. However, if you add a reasonably good DAC to the PC... or better yet, use a TacT 2.2x as a DAC and room correction, you will be able to easily compete with the Bel Canto.

What I heard in a direct comparison of a BC 7,k CDP and my Audio PC was that the BC did technically flesh out more detail... but still sounded like a digital source... the PC sounds more analog to me.

Comparing my PC to one with the Lynx (Hantra's), I noted the sound on mine was a little thinner... but my card is much cheaper... an M-Audio Delta DIO. I plan on using an external DAC down the road...

-C

WerTicus · « **Reply #46 on:** 3 May 2005, 08:04 pm »

yes well i would expect the canto to win too - but im hoping the sound coming from the l22 is going to be 'satisfactory' and that i dont have to add any more components. i certainly like the idea of having my pc go straight to my amps...

anyways l22 is here on monday!

WerTicus · « **Reply #47 on:** 9 May 2005, 02:07 pm »

i got my lynx l22 - i need the pin out diagram for the breakout cables... if anyone has it handy!

thanks!

EchiDna · « **Reply #48 on:** 6 Jun 2005, 11:11 am »

hmmm the lynx must be good werticus... no updates for a while

love to know what you have made of it!

cheers,

WerTicus · « **Reply #49 on:** 6 Jun 2005, 12:34 pm »

ehehhehe i am listening to it right now.

its very nice actually

but im not using it in balanced mode atm because i am still waiting for my balanced nuforce amps!

there is nothing wrong with the lynx however

i will write a review of the nuforce the lynx and the reference 1's all at the same time i think in a couple weeks! with pix!

Jon L · « **Reply #50 on:** 6 Jun 2005, 03:38 pm »

Quote from: WerTicus

ehehhehe i am listening to it right now.

its very nice actually

but im not using it in balanced mode atm because i am still waiting for my balanced nuforce amps!

there is nothing wrong with the lynx however

i will write a review of the nuforce the lynx and the reference 1's all at the same time i think in a couple weeks! with pix!

I would not write a review without getting rid of the Lynx breakout cable, which sounds sub-par, to put it mildly. On page 31 of Lynx manual, you can find the DB25 connector pinouts.
http://www.lynxstudio.com/manuals/LynxL22Manual.pdf

Get a DB25 connector and terminate it with XLR connectors with couple inches of good wire. This way you can use the XLR interconnect of your choice, which makes a big difference.

Depending on the overall system synergy, Foobar in ASIO with SRC resampler (not ssrc) resampling to 176.4kHz into Lynx can be worth trying instead of straight 44.1kHz..

WerTicus · « **Reply #51 on:** 6 Jun 2005, 03:49 pm »

i'm going to be running fully balanced connections next week when i get my XLR nuforce amps.

and i am building the interconnects for this as we speak! got some high quality wire from pat for this job (war audio)

I'm not sure what your talking about with the upsampling, can you desribe exactly where this setting is? i can see SSRC but not ASIO?

Lynx does not seem to support kernel streaming? also i thought that was relevant to digital out only?

Jon L · « **Reply #52 on:** 6 Jun 2005, 05:41 pm »

Quote from: WerTicus

i'm going to be running fully balanced connections next week when i get my XLR nuforce amps.

and i am building the interconnects for this as we speak! got some high quality wire from pat for this job (war audio)

I'm not sure what your talking about with the upsampling, can you desribe exactly where this setting is? i can see SSRC but not ASIO?

Lynx does not seem to support kernel streaming? also i thought that was relevant to digital out only?

You have to download Foo output ASIO (dll) plugin from here http://www3.cypress.ne.jp/otachan/

and drop it into "components" folder in C:/ programs->Foobar2K

Then open Foobar->Output. You should now see a choice called ASIO, which has its own choices in pull-down, which should include your Lynx card. Make sure to set output to 32 bit (for Lynx). You might as well download from that site the ASIOCaps utility, too, which will let you increase ASIO buffer size, which you absolutely need if upsampling in ASIO.

Without ASIO (or KS), your foobar output is currently being resampled automatically to 48kHz by the dreaded K-mixer, and yes ASIO sounds significantly more transparent.

KS is also supported by Foobar and Lynx, BUT to me it doesn't sound as clear and detailed as ASIO for some unexplained reason. You can find the Kernel Streaming plugin by doing some googling.

As far as SRC resampler and other computer PC needs, Steve has some nice links and explanations on his site under "computer audio" at empiricalaudio.com..

WerTicus · « **Reply #53 on:** 6 Jun 2005, 06:57 pm »

yup your right asio is more detailed.

but upsampling asio seems to do nothing.

also it does not seem to take much cpu, but i have a beast here (dual opteron)

to upsample asio do i still use SSRC or is it done some other way?

Jon L · « **Reply #54 on:** 6 Jun 2005, 08:02 pm »

Make sure the resampler is placed in the left column in "DSP" in Foobar. You have to click twice on all the choices on right column to move it to left. Otherwise, it's not active.

SSRC comes with foobar, which I'm not a huge fan of, but you can use SRC (Secret Rabbit Code) resampler from here http://pelit.koillismaa.fi/plugins/dsp.php

All the resamplers only list up to 96kHz, but you can actually manually type in 176400 or 192000.

The difference between 44.1 and 176400/192000 should NOT be subtle; if it is, there is something not working right.

Also, once again, I strongly recommend downloading that AsioCaps utility to set buffer to max to prevent skipping w/ upsampling. Reducing the Foobar thread priority bar by one click is almost mandatory as well.

WerTicus · « **Reply #55 on:** 7 Jun 2005, 07:06 am »

everything seems to be working thanks for your help jon

i assume everything is internally 32bit with this sound card? (yes i have play back at 32bit fixed point)

now just to figure out how to get DVDA files into a wav or something