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iFi Audio Nano iDSD discussion + impression - Page 17

post #241 of 1720
If under windows it won't work then which device I need to buy? Would iPod be sufficient?
post #242 of 1720
Quote:
Originally Posted by kugino View Post
 

not intending to incite any mac/windows war, but wow, can windows make it any more difficult to output digital audio? 

I have to admit - I'm a computer programmer - I'm also getting into my late 50's and I have grown TIRED of how UN-userfriendly windows is. When I was young it was a challenge. NOW - I cant stand it. I feel there is Apple in the future. We already have several iPhones and an iPad in our home.... I think there will be more. - how about an iPad with some USB outs etc?

post #243 of 1720
Quote:
Originally Posted by koolas View Post

If under windows it won't work then which device I need to buy? Would iPod be sufficient?

i'm pretty sure there is a way to output DSD in windows. some guys here might be able to help...i'm not help - i use a mac.

 

Quote:
Originally Posted by earwaxxer View Post
 

I have to admit - I'm a computer programmer - I'm also getting into my late 50's and I have grown TIRED of how UN-userfriendly windows is. When I was young it was a challenge. NOW - I cant stand it. I feel there is Apple in the future. We already have several iPhones and an iPad in our home.... I think there will be more. - how about an iPad with some USB outs etc?

don't hold your breath. all you'll ever get on the ipad is the lightning in/output. but using the camera connection kit isn't a bad way to output audio on the iDevice. can play DSD files to the iDSD using the CCK and appropriate app like the onkyo HF Player.

post #244 of 1720
Ok, good news. With JPlay I was able to play DSD64 and DSD128 smily_headphones1.gif
On Linux it apparently does work. I don't know why would it not work on my laptop, but on desktop both Pulse Audio and Jack work fine.
I might try again with laptop. I would also prefer to make Foobar work rather than using JPlay.

BTW SQ of DSD files played with JPlay really astonishing:)
post #245 of 1720
Quote:
Originally Posted by iFi audio View Post
 

 

Hi,

 

We use an Oversampling filter for PCM at sample rates lower than 352.8/384 KHz, but no "Upsampling".

 

thank you

 

Different methods, same result!

post #246 of 1720
Quote:
Originally Posted by marcie View Post

Different methods, same result!

Well there is slight difference. The up-sampling is any type of sample rate conversion from lower to higher, while oversampling is one type of such conversion - AFAIK oversampling it is ZOH method, which produces square samples. It is like viewing low-rez image on high-res screen. If low-rez imge is half the pixels of the high-rez screen then you can use ZOH, that is every pixel of low rez image is shown using two adjacent pixels of high rez image. The effect is as your display was low rez. Other up-sampling techniques involve linear filter, or cubic filter, and even sinc filter can be used. However most of you probably noticed that linear filter used to scale low rez images makes images appear blurred. Situation is much better if you use cubic or sinc filter. However sharpest image you get with ZOH. I which LCD screens had an option to enforce ZOH...

So, it means that lower rez music will sound digital on iFi unless you lock output to DXD. I run Jack at 352.8kHz/24bit under Linux and use Audacious to play music. The white LED on iFi confirms that I'm sending 352.8kHz data over USB, and the sample rate conversion is done in software. And in that software I may choose to use ZOH, linear, or sinc if I wish.
post #247 of 1720

Hmmm. The benefits of upsampling vs oversampling are greatly debated, and presumably overstated. See here for a technical description:

http://www.audioholics.com/audio-technologies/upsampling-vs-oversampling-for-digital-audio

 

Many other articles are available on the net, but those not providing technical descriptions should be avoided. I recall reading statements from the likes of John Westlake and Paul Miller commenting on the "sameness" of results. It's a minor point, but I'm a bit wary of manufacturers claiming benefits where really, there are none. That's why I made my first comment on 'different method, same results'. It takes absolutely nothing away from the iDSD Nano, which I believe is a tremendous achievement!

post #248 of 1720

Hi

 

This comes from AMR's Chief Designer:

 

Oversampling is not automatically ZOH, in FACT, ZOH would be pointless in an oversampling filter, as it, in effect does not " oversample" nor "filter" at all, might as well use Non-Oversampling.

 

However Up-conversion to 352.8KHz (for 44.1/882/176.4 KHz Sample rate recordngs) or 384KHz (for 48/96/192KHz Sample rate recordings) with ZOH can be used to in effect place the iDSD into what amounts to non-oversampling mode. The downside is that ZOH attenuates the HF response of the resulting signal, for example CD-Standard PCM audio will face 3.2dB attenuation at 20KHz starting to rolloff as low as 5KHz - this can make the music sound too dull.

 

Linear interpolation is equally pointless, it does provide some mild filtering but again does not really help, but it produces even more HF roll-off than ZOH.

 

Modern Digital filters use different structures (tapped delay lines with multiple weights to mix together the different delayed outputs - this is somewhat reminiscent to the old style of analogue tapped delay line chroma filters in old Colour TV's). This rejects ultrasonic images depending on filter slope. 

 

The "standard filter" on the iDSD in effect performs "sinc" type interpolation, which has the best "on paper" performance, with completely flat frequency and phase response up to half the sample rate and then a very steep rolloff. It also results in a symmetrical impulse response with filter ringing BEFORE and AFTER the main impulse. 

 

However many listeners tend to associate this type of filter with "digital sound", for reasons that so far lack any reliable proof, though many theories to the "why" exist. 

 

The iDSD offers an alternative filter that is generally evaluated as more "analoge" or "musical" than "best sinc". This alternative minimum phase filter on the iDSD is a different animal to anything discussed so far. This type of filter was pioneered by Wadia as "Digimaster" and Pioneer as "Legato Link" and has since been picked up by a number of manufacturers. 

 

These filters do not attempt perfect response flatness and allow some HF attenuation at half the sample rate (but much less than ZOH, so the "dull sound" effect does not happen). And they are less steep, so some ultrasonic images are allowed through (though much less than ZOH) but the impulse response lacks pre-ringing. 

 

We find these filters a good compromise between filtering ultrasonic noise and time domain fidelity vs. amplitude response flatness. The type of filter embodied by the iDSD's minimum phase filter is generally not available in resamplers such as those based on SOX (most common in playback software).

 

Additionally the digital filter in the iDSD is adaptive, in terms of sample rate. The higher the source sample rate, the less the oversampling ratio. So for example CD standard PCM is oversampled eight Times, but 192KHz audio is oversampled only two times, in effect the oversampling filter always performs integer ratio oversampling to 352.8KHz or 384KHz with either sinc or minimum phase filtering/interpolation (selectable).

 

We recommend to leave the digital processing to the built-in digital filter in Minimum Phase mode, though of course our customers are free to choose whatever way they wish, however we would strongly recommend to try the internal minimum phase filter FIRST, before trying other options.

post #249 of 1720
Sounds like iDSD is super-awesome beast biggrin.gif

I've got one question: which position of the switch is which filter, because labels say "MINIMUM STANDARD" and "FILTER". I use the first one, but honestly I don't think I can hear difference between those two. Maybe need to do more listening.

However... I've finally got Foobar to play DSD as DSD - yes LED goes magenta. But once I use foo_output_dsd, then all the CD music is converted by Foobar into DSD128 format, which I find good. I think the PCM converted to DSD sounds better than oversampled PCM. I think the oversampled PCM tended to sound digital, i.e. I think hear ringing and aliasing artifacts (Yet need to compare both positions of filter switch). But when Foobar converts PCM to DSD the result is just amazing. The whole thing about DSD is not about wider frequency range, because human ears can't hear above 20kHz, but it is all about smoothness. Correct me if I'm wrong, but with DSD you will never have any aliasing artifacts, no ringing, DSD really sounds like it is digital Vinyl and iDSD does magic, true magic...

The only problem I have with iDSD is its reliability. When Foobar switches from DSD64 to DSD128 it fails sometimes, I had issue that there was no sound as well. On Linux sometimes in the middle of song it can stop playing and I have to restart Jack. Actually I haven't experienced that problem with Jack at 382.8, so if that is the solution I'm OK with that. It sounds like I wouldn't be using built-in oversampling, but it's good to know its engineered so well. Sinc filter - who would expect - I'm really astonished smily_headphones1.gif
post #250 of 1720

I listen to rock/pop on ripped CDs at home and 320 MP3s at the gym.  I doubt I will buy any DSD files soon.  Listened this morning with JRiver changing the output format using "DSP & output formats" under Options/settings.  I used Brainwavz B2s and am on the road with just a few listening choices.

 

"DSD in DoP format" (blue light on iDSD: DSD 2.8)  was clearly worse than no encoding (sending unchanged 44.1kHz > green light on iDSD) with rolled off top end and muddy bass - disappointment.

 

"2xDSD in DoP format" (magenta light on iDSD: DSD 5.6) was slightly better than no encoding.  Voice and guitar were a bit more natural and bass guitar and kick drum were more distinct.  I would probably be happy with either option.  Unprocessed 44.1 may have been a bit forward in the mid-range, though this may be an unfair/inaccurate comment.  YMMV.

 

2xDSD in native format didn't work on my laptop,  Perhaps it lacks power with a JRiver benchmark of 2152.

 

The performance is very strong with 16/44.1 material and it seems fairly priced.  I will compare it to a Ross Martin bare beast later in the week (which I have liked the past couple of years and have at home) .

 

FWIW, I was formerly julius_the_cat on this forum, but my previous username was disabled. I find the technical part of music less interesting than it was to me a few years ago.

post #251 of 1720
Quote:
Originally Posted by juliusthecat View Post

I listen to rock/pop on ripped CDs at home and 320 MP3s at the gym.  I doubt I will buy any DSD files soon.  Listened this morning with JRiver changing the output format using "DSP & output formats" under Options/settings.  I used Brainwavz B2s and am on the road with just a few listening choices.

"DSD in DoP format" (blue light on iDSD: DSD 2.8)  was clearly worse than no encoding (sending unchanged 44.1kHz > green light on iDSD) with rolled off top end and muddy bass - disappointment.

"2xDSD in DoP format" (magenta light on iDSD: DSD 5.6) was slightly better than no encoding.  Voice and guitar were a bit more natural and bass guitar and kick drum were more distinct.  I would probably be happy with either option.  Unprocessed 44.1 may have been a bit forward in the mid-range, though this may be an unfair/inaccurate comment.  YMMV.

2xDSD in native format didn't work on my laptop,  Perhaps it lacks power with a JRiver benchmark of 2152.

The performance is very strong with 16/44.1 material and it seems fairly priced.  I will compare it to a Ross Martin bare beast later in the week (which I have liked the past couple of years and have at home) .

FWIW, I was formerly julius_the_cat on this forum, but my previous username was disabled. I find the technical part of music less interesting than it was to me a few years ago.
Doesn't surprise me that you'd hear no sonic improvement via software upsampling...
post #252 of 1720
Quote:
Originally Posted by koolas View Post

Sounds like iDSD is super-awesome beast biggrin.gif

I've got one question: which position of the switch is which filter, because labels say "MINIMUM STANDARD" and "FILTER". I use the first one, but honestly I don't think I can hear difference between those two. Maybe need to do more listening.
 

 

There are 2 filter options:

 

"Minimum Phase Filter" OR "Standard Filter"

 

The switch in the upper position is "Minimum Phase Filter."

 

The switch in the lower position is "Standard Filter."

 

Hence "Minimum/Standard" and "Filter."

 

 

 

If you try a piano concerto track, a REAL piano sounds "woody". A piano done not so well sounds "glassy."

 

To us, Minimum Phase sounds more like the former.

post #253 of 1720
Quote:
Originally Posted by koolas View Post

When Foobar switches from DSD64 to DSD128 it fails sometimes, I had issue that there was no sound as well. On Linux sometimes in the middle of song it can stop playing and I have to restart Jack. Actually I haven't experienced that problem with Jack at 382.8, so if that is the solution I'm OK with that. It sounds like I wouldn't be using built-in oversampling, but it's good to know its engineered so well. Sinc filter - who would expect - I'm really astonished smily_headphones1.gif

 

Drop us a PM or email.

 

We'll run through the Foobar settings with you.

 

We can only offer suggestions. As you know, Foobar is open source - hence not expected to be super-stable.

 

JRMC and AN+ are what we recommend and they are not expensive yet are reliable.

 

If you try JRMC (trial) you will find the iDSD does what it says on the tin!

 

thank you


Edited by iFi audio - 2/16/14 at 4:42am
post #254 of 1720
Hi all, sometime back, someone asked me to compare the iDSD to X3.

From what I can hear from FLAC files sourced from both RBCD and a few high res (192/24). I didn't compare to DSD files as the X3 cannot play them. The differences are:
- iDSD has a wider, deeper sound stage
- iDSD has more fine details and "air" especially for instrumentation. Vocally, they are almost the same, edge goes to iDSD
- On X3, the bass, especially drums is quite "dense" that comes across as heavy and solid. On iDSD, it is slightly lighter but has some air in it. Kinda like a heavy/dense vs light gnocchi, if you know what I mean. Given the reputation of the X3 of being "warm" it is not surprising. Not that the iDSD a lot less bass, but a lot more moderate is the feel that I get.
- on iDSD, the violin and other string instruments somehow sounds sweeter, more air and has a sharp edge to it. It is more exciting and enjoyable when compared to the X3 where the strings are rather heavier and "rounded" and very ordinary.

On PC: Foobar 2k + foo_dsd_asio ->iUSB -> Gemini cable -> iDSD -> Shure 846 (default filters)
post #255 of 1720

Ok, I've encountered another issue with the iDSD when I connect it to my other laptop.  On a USB 3.0 port, I can't play DSD files (DFF, DSF) files properly.  PCM has no issue, but on my i7 Lenovo Helix, when I plug the iDSD with the USB cable that comes with it, using foobar, hqplayer or JRMC, all the DSD files sounded staccato and choppy.  I tried both ASIO and WASAPI drivers with the same outcome.  The ASIO drivers even hung the players and seems to even wait for kernel events (the program UI closes, but the process is still visible in the task manager).  I tried different ports, linking to and using the USB 3.0 port on my monitor and switching cables but to no avail.

 

What worked was when I switched in the BIOS for the USB ports to behave as an USB 2.0 port.  The other way was to hook up an USB 2.0 hub and plug the iDSD to it.

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