Peter

Sorry, but I did not express myself precisely. What I meant was I guess 09-z2 will prevent an attempt to 2x upsample a 24-192 file, if DAC max capability is 24-192 (as I tried by mistake).
I appreciate the 24 bit upsampling capability and am following with interest the evaluation of the sound quality btw 24-48 with 4x upsamping, 24-96 file with 2x upsampling and native 24-192 from forum members with more sophisticated systems than mine.
 
Bjorn

Peter

Using 09-z1 I tested to upsample a 24-96 file with 2x AP (24-192 DAC) and played OK- good sound.
I then accidentally played a 24-192 file with setting still at 2x AP and 24-192 file played, but at half the recorded speed.
I guess this will be prevented in 09-z2 as an unsupported output.

Thanks for all your good work

Bjorn 

Peter

In trying to get my son's computer (AMD, 2GB memory, Realtek sound on MSI motherboard) I had to clear old Config file to get it going after upgrade to 0.9Z.
I still have a problem with WPChange notices for each new file played, so may have to go through clear old COnfig file once more.
However, once running I tried different settings for KS sound quality - Adaptive gave good sound with large buffer (2048) and Q1=1.
I then tried Special mode, but had intermittent clicks and went back to play with Adaptive 2048 buffer setting.
To my surprise, I could this time lower the buffer to 384, and still get good sound, which was not possible before switch to play in Special mode.
However, it was not until later I noticed that by trying out Special mode, Q1 had been set to 14 by Specail mode, and remained at Q1=14 even after I returned to Adaptive mode.
In other words, in Adpative mode buffer=384 was obtained in combination with Q1=14.
Question: Is this generic, or only happening to me. I thought Q1 would return to its previous value when I returned to play in Adpative mode, or does this require that I save the Adpative mode settings somehow and return to this preset if I want to specifiically associate Adaptive mode with Q1=1.

Bjorn

Peter

Great sound from 0.9z, but for now 0.9z only operating with Engine 4 (KS).
My laptop has built-in HD sound card that only seems to want to work with Engine 3 (non KS), up to 192-24.
I can listen by plugging in Logitech ear phones, but then limited to 48-16.
Is the plan to make 0.9z (or future version) work with Engine 3?

Best regards

Bjorn
 

Peter

Converting vinyl to digital and digital RIAA filter

1. Digital RIAA filter plug-in
see, free download (VST)
http://www.kvraudio.com/get/3550.html

2. Saving files
My thought was to record and save vinyl as 96/24 files.
As for upsampling in XX, I suppose upsampling of 96/24 files will require a 32 bit DAC, such as NOS1 (similar to upsampling of 16 bit files requires a 24 bit DAC).

Bjorn

Peter

As you invited comments, let me expand on the issue of playing analog (vinyl LPs) and RIAA conversion through XX software and NOS1 DAC.

First, my thoughts to tranfer my vinyl collection (vinyl rip to 96/24 and save files to disc) was more driven by the convenience factor; with vinyl in digital domain benefits would be
a. stored in central digital music library and easily accesible (togteher with CD rips and digital dowloads) via computer network to play at any location in my home
b. ability to play vinyl tracks in any order (from one or several LPs)
c. possibility to play through XX and take advantage of its superior abilities

Thus, the above driving the decision to look into option to transfer vinyl into the digital domain, and the discussion of better to apply RIAA in the analog or digital domain stems from that.
In other words, transfer vinyl to digital has benefits over and above the advantage of possibly applying RIAA filtering in the digital domain.

I think useful to outline three different categories of LP listening/ digital conversion using XX and NOS DAC

1. Playing LPs (in real time) with ADC conversion in NOS1 and applying (real time) filtering in XX.
  The reason for vinyl analog to digital conversion is presumably to take advantage of XX digital filtering capabilities; if not for RIAA filtering, then room adjustment and/or x-over filtering. In fact, I see no real separation between digital room adjustment filtering and RIAA filtering. In other words, would it not be feasible for a system set up to use a digital room adjustment filter in XX to have two separate filters – one room adjustment filter for regular digital (CD) files and the second the same room adjustment filter with the RIAA curve imbedded (rather than two separate filters) for vinyl playback.
Also would be great to have NOS1 capability to pipe out the signal both in digital and analog simultaneously; i.e. while playing vinyl and listening to the NOS1 analog OUT signal, the NOS1 digital OUT signal could simultaneously be routed back to the computer and saved. This way, vinyl ripping would not be a separate chore, but a by-product of listening to LPs (optimal if option provided to save the digital file either with digital RIAA filtering imbedded or w/out RIAA adjustment, see below).     

2. Playing a digital file converted from vinyl, but stored on disc as a RAW file without imbedded RIAA filtering.
The file would be subject to digital RIAA filtering in XX when played. However, as the RIAA is not embedded in the file before storing on disc, playback of the file would benefit from continued improvements in XX digital RIAA filter capabilities. (Similar idea to RAW file format for photos).  Playback would require XX (for RIAA filtering), but not necessarily require use of NOS1 DAC.

3.  Playing a digital file converted from vinyl, and stored on disc as a file with RIAA adjustments imbedded (as a result of either analog or digital RIAA filtering).
As for any other digital file neither use of XX or NOS1 DAC strictly required.
However, would be great to have XX capability to upsample a stored 96/24 file to 192/24 and 384/24, the latter presumably requiring NOS1 DAC.

Bjorn

 

Peter

The thread has earlier touched on the possibility of digital filtering (in software) as well as the possibility for the DAC to perform the ADC of the analog signal from a turntable.

From that perspective, I am interested in your thoughts on the pros and cons of applying the RIAA filtering in the analog vs digital domain.
From a practical perspective, I assume that for implementation of digital RIAA filtering  in XX software, analog signal (from turntable w-out analog phono stage) would first need to be fed to the DAC for ADC conversion, then digital signal fed back to computer for RIAA filtering in XX software and finally digital signal fed back to the DAC for conversion to analog before signal OUT fed to the amp.

In general, if analog signal from the turntable (w/out analog phono pre-amp) is too low for processing in your DAC and some form of phono pre-amp required between the turntable and the DAC, analog RIAA filtering may just as well be performed in the phono pre-amp (and forget about RIAA filtering in the digital domain).

Best regards

Bjorn

Peter

When you suggested covering the port in a vented speaker, do you mean completly sealing (as with a piece of solid wood) or only partially sealing (as stuffing the port with some foam, as I have seen suggested on a a speaker construction site).

Bjorn

Peter

You have stated that sufficient speed in the amplifier is essential for excellent sound.

I am just wondering if running a stero amp in balanced (mono) mode is a way to increase speed, or this just adds power to the amplifier whithout any impact on the speed.

Bjorn

Peter

Thanks for your explanation - provides a good outline of technical side of the operation of (earlier and latest version) NOS1.

Let me just state my understanding of these operating principals and how you intend to implement those in the NOS1, and you can correct me if I have misunderstood.

1. Switching btw Balanced and SE operation.
The switching of internal operating mode of NOS1 btw Balanced and SE. In other words, not a switching btw Balanced vs SE digital input.
In effect, the DAC does not even need a Balanced input as as a balanced +/- digital signal can be generated in the DAC from a digital SE(+) IN signal.

2. Balanced vs SE output
Balanced vs SE output is linked to the selection of operating mode; i.e. Balanced (XLR) output only active when Balanced operating mode selected and SE output only active when SE operating mode selected.

Bjorn

Peter

In relation to measuring the noise in your DAC project you stated that noise would be (at least 3 dB) better if measured with balanced lines.
I take that to mean same signal from the DAC's digital section, but processed through the balanced section of the analog stage in your DAC, rather than processed through the unbalanced section and measuerd at the unbalanced analog output (as in your posted measurement pictures).

In other words, you are not speaking about the difference in the DAC's performance when fed a balanced source (digital IN) signal versus an unbalanced source (digital IN) signal. 

Would be interersting to hear your thoughts on feeding unbalanced versus balanced digital source signals to an external DAC - in general as well as related to your DAC project (listening and mesurement findings).

Best regards

Bjorn

Peter

Unable to find any Phasure sub-directory or a Config file, as XX never loaded.
However, Realtek Audio is showing problems (even with updated driver); Realtek sound OK in WIN Media Player, but does NOT play music on HDTracks on the Net.
With USB Logitech earphones, I have sound both in WIN Media player and to play music samples on HDTracks on the Net.
For that reason , I have now disabled Realtek and using Logitech USB earphones for sound (at least until XX can be loaded)

So, thankful for direction re your mention of dot-net possibly not properly installed.
FYI, XPSP3 newly re-installed after a HD malfunction, thus very possible some essential XX support functions not (yet) properly installed.

Best regards

Bjorn 

Peter

I tried installing XX0.9y-08 on my son's PC running XPSP3 (AMD Athlon 3000+, 2GBRAM, 150GB IDE HD), but got an error message
"The application failed to initialize properly (0xc0000135). Click OK to terminate the application".

I thought problem possibly with XX0.9y-08/ XX0.9y-08c, but same problem with XX0.9y-06 and XX0.9y-04, which leads me to think something may be wrong with the PC (memory?).
Sound on motherboard is Realtek AC97 Audio, but it also failed (same error message) when trying with Logitech USB headphones that work fine with XX on my main Vista64 PC.
Would appreciate your assitance to diagnose and suggestion how to fix.
For a start, I do not really know if the error message is a WIN XP message or a XX message.

No panic with this - not my intention to prevent you from enjoying an Easter break.

Happy Easter
Bjorn   
 

Peter

My remarks for info only, as you had invited commenst about user experience.
For serious listening at home I will use external DAC (M-AUdio Profire 610 via FW); had to get laptop FW Expresscard (with TI chip set) for optimal sound, laptop internal FW did not connect well with M-Audio FW DAC. 
When away from home and using laptop internal sound card,I agree  Engine 3 best available option.

Best regards

BJorn

Peter

Unfortunately, no KS on Acer Travelmate 8204 laptop (4MB RAM, Vista 32x) using the internal soundcard. According to spec this is a 32 bit Intel HD spec. Unable to get more detail on the soundcard/chip, but it plays 192/24 HIRES material (Linn download piano test file) - at least of first, later often stumbles with note re. insufficient memory to allocate. Similarly, can play 44.1/16 material in Quad (Anti-image), IF set to 96/192 kHZ only.

On the other hand, KS available (16 bits only), when connecting Logitech headphones via USB.

Despite the above setback (prompting me to use Acer laptop with external DAC), many thanks to you for making excellent sound of KS available under XP (works OK on my old laptop and on my son's XP desktop computer).

All the best

Bjorn