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USB to SPDIF converters shoot-out : EMU 0404 USB vs. Musiland Monitor 01 USD vs. Teralink-x vs. M2Tech hiFace - Page 57  

post #841 of 1712
Quote:
Originally Posted by slim.a View Post
By the way, if you want to clear the bass/warmth issue with the Teralink-X2, you might want to try the Wireworld Ultraviolet USB cable. In my case, it has cleared the mid bass bloat and excessive warmth of the older Teralink-X.
I doubt the cable would rectify the issue to the extent I would like. The difference between the HiFace's bass and the Teralink's is not subtle. I even preferred my motherboard's bass over the Teralink's. Using a few different drivers, I think there is potential for the issue to be fixed through software. I'm particularly interested in Fujak's Tenor ASIO driver.
post #842 of 1712
Anyone know how this M2Tech hiface sound compare to X-Fi Surround 5.1 USB soundcard?
I mostly play MP3/Flac/Wav music files(16bit/44.1khz.).
post #843 of 1712
Thread Starter 
Quote:
Originally Posted by Shahrose View Post
I doubt the cable would rectify the issue to the extent I would like. The difference between the HiFace's bass and the Teralink's is not subtle. I even preferred my motherboard's bass over the Teralink's. Using a few different drivers, I think there is potential for the issue to be fixed through software. I'm particularly interested in Fujak's Tenor ASIO driver.
In the case of the Teralink-X the changes in drivers (stock, ploytec asio, cmedia) were much more subtle in nature in comparison with a change in usb cable (regular vs. Wireworld Ultraviolet USB).
post #844 of 1712
Thread Starter 
Quote:
Originally Posted by duckymcse View Post
Anyone know how this M2Tech hiface sound compare to X-Fi Surround 5.1 USB soundcard?
I mostly play MP3/Flac/Wav music files(16bit/44.1khz.).
The M2Tech hiface is a digital to digital (usb to spdif converter) while the X-Fi Surround 5.1 USB seems to be more a usb soundcard with after thought optical output. Do you use your X-Fi with an external DAC? If so, and if you DAC accepts coaxial, you will most probably get an improvement in playback performance (bit perfect, lower jitter, ...)
post #845 of 1712
Interesting thing about this discussion is that typically in a DAC the reciever has a PLL that virtually eliminates Jitter above a certain frequency. So the difference between the Hiface & TerralinX 's bass is a true support that Jitter matters. Still not sure which of the two has better bass based on the descriptions (grados like warm bass). Also would be interesting to know how battery or bench PS clocks on the Hiface affects the bass. This thread is really getting cutting edge with the open honest impressions being shared. I am really commend you all by keeping this thread away from the confromtational/defensive spiral that a lot of these turn into.
post #846 of 1712
Thread Starter 
Quote:
Originally Posted by regal View Post
Interesting thing about this discussion is that typically in a DAC the reciever has a PLL that virtually eliminates Jitter above a certain frequency. So the difference between the Hiface & TerralinX 's bass is a true support that Jitter matters. Still not sure which of the two has better bass based on the descriptions (grados like warm bass). Also would be interesting to know how batter or bench PS clocks on the Hiface affects the base. This thread is really getting cutting edge with the open honest impressions being shared. I am really commend you all by keeping this thread away from the confromtational/defensive spiral that a lot of these turn into.
Yes, it seems that the difference lies in the low frequency noise/jitter of both transports. Using better clocks and better power supply filtration would probably boost the performance of the hiface.
Marco from m2tech told me a few weeks ago that he was working on a "super-converter" but he didn't give much information.
One thing to keep in mind when comparing the hiface vs. Teralink. The Hiface is async and uses 2 true audio clocks (for multiples of 44 and 48 frequencies). So its performance is mostly dependant on the quality of the clocks (and power supply).
The Teralink is an adaptive unit (not async) that uses one very high quality clock. But 44.1 has to be derived from a multiple of 48. So its performance will depend on the computer, the clock, the PLL that generated the 44.1 frequency and the power supply (and the quality of the usb cable).
post #847 of 1712
Quote:
Originally Posted by slim.a View Post
The Teralink is an adaptive unit (not async) that uses one very high quality clock. But 44.1 has to be derived from a multiple of 48. So its performance will depend on the computer, the clock, the PLL that generated the 44.1 frequency and the power supply (and the quality of the usb cable).
Yes I wish I was digital savy enough to put a 44.1 multiple clock in my TerraLinX, I think that would be tough to beat.
post #848 of 1712

Slim appreciation.

Quote:
Originally Posted by FauDrei View Post
Slim, do you ever eat, sleep and do other stuff beside evaluating and comparing audio gear?

I mean, the sheer amount of quality information that you compile on this board (plus some subjective opinions)... great stuff man.

Hope you do not ignore the rest of your life because of this. <gives a pat pat on the shoulder>
X2

I run Apple Macbook pro 15 / itunes / Amarra / Hiface bnc with the Stereovox VH2 cable.

I decided on the hiface based on this thread and confirmed by the 6 moons review.

The sound is awesome, bass is great and unused lossless songs sound great, even Itunes downloads ( bought for investigation ) are more than acceptable.

I have no issues with the A GD Ref 1 , and Bel Canto dacs but get dropouts with the Valab nos dac, funny though as 1st evening was sublime considering the cost of the dac, subsequently serious dropouts with Valab only.

The Valab does not accept Amarra at all but Itunes / hiface ok . Any thoughts?

Oyaide db 510 bnc on the way.

Anyhow, a big thanks to Slim and other posters.

Hishade.
post #849 of 1712
Thread Starter 
Thanks for the appreciation guys
post #850 of 1712
Quote:
Originally Posted by slim.a View Post
Marco from m2tech told me a few weeks ago that he was working on a "super-converter" but he didn't give much information.
I wonder how much the new converter's gonna cost, and if it'll affect the existing price of the hiface
post #851 of 1712
Thread Starter 
Quote:
Originally Posted by noinimod View Post
I wonder how much the new converter's gonna cost, and if it'll affect the existing price of the hiface
I don't think we can expect a lowering of the price of the hiface. Since it is fairly priced against the competition and is selling well, I don't see why they would change their price . The new converter is likely to cost a lot more than the hiface if they decide to go with an external enclosure ... But that is only speculation for now
post #852 of 1712
Thread Starter 

Digital Filters, Minimum Phase and Upsampling:

Digital Filters, Minimum Phase and Upsampling:

While I have already mentioned digital filters and upsampling in separate posts, I thought it could be nice to compile some of that in a single post. So here are my findings on the subject: nothing new, just a compilation of thoughts.


Digital filters:

One of the discriminating factors between different DACs is the digital filters. Most modern chips include a built-in digital filter (for cost savings reasons) but they perform very poorly in general. But how much difference does it really make?
In my current DAC, I have the possibility to swap between 2 digital filters: the PMD100, which is a HDCD capable digital filter made by Pacific Microsonics, and the DF1704 digital filter made by TI. The two digital filters have 2 distinct sonic signatures (for those interested in the details, they can read my review of the audio-gd dac19mk3 where other users have reported similar differences between the 2 filters).
When playing 16/44 data, the DF1704 is fast and analytical, slightly on the bright side of neutral and the PMD100 is warmer and more “analogue” like.
While the PDM100 is supposed to be limited to 24/55, it seems to work at 24/88 (but not 24/96).
Well, by playing (native) 24/88 data on both, they sounded closer than they did at 16/44. The DF1704 smoothed out and the PMD100 had a little bit more sizzle on top.

That was similar to what I have experienced with some sigma-delta DACs that I have found to sound harsh at 16/44. Usually they sound much better with 24/96 data or upsampling to 24/96. The reason behind the perceived improvement is not due to the fact that we hear the extra data. It is rather due to the fact that we do not have to hear as much the nastiness of the digital filters operating close to our audibility range.

On many digital filters whether they are outboard filters (such as the DF1704) or built-in into the dac chip (such as the CS4398) the designer has sometimes the choice between a sharp/fast roll-off and a slow roll-off.
The sharp roll-off usually gives a flat measured frequency response on the 20-20,000Hz range, which looks good for published measurements. But that flat frequency response up to 20K comes at the expense of phase performance (which has an impact on soundstaging among other things).
The slow roll-off on the other hand is down from 1 to 3 db (in general) at 20 khz but has a better phase linearity, which is more audible to the human ear.
So when given the choice between what looks good on paper (a flat frequency response) and what sounds nice to the ear (phase response), most designers choose the first option.
I have looked at the measurement of many DACs on the stereophile website and some of the most expensive and better sounding DACs seem to have that roll-off on the highs. Weirdly enough most budget DACs and soundcards seem to have a flatter high frequency response. I am not saying that we shouldn’t pursue a flat high frequency response. But I am just saying that when you see a budget DAC that has a flat response for 16/44 data beyond 20 kHz, and if the DAC doesn’t use a fancy DSP, it is most probable that it has sacrificed the phase response for the good looking Frequency response measurements.
By the way, if you are still thinking that a flat frequency response is absolutely necessary to have a realistic sound, just think about the following.
Let’s suppose you listen at a violin in very reverberant room, then you listen to a violin in very damped room of a different size from a different distance. Then listen to a violin in a speaker system that has been equalized to have a flat frequency response. As far as I am concerned, while situation 1 and 2 will have different Frequency Response measurements, my brain will analyze and detect the sound as being real and live. When I will listen to situation 3, I most probably won’t be fooled thinking that the violin is real.

Fortunately, there is a way to go partially past those poor digital filters. By upsampling data to 24/96, you minimize the audible effects of those filters, as any aliasing/distortion they generate will be pushed further up in frequency range.


Impulse response & Minimum Phase Filter:

While I have mentioned the Frequency response and phase response of digital filters aspects, I haven’t mentioned yet the impulse response.

When compared to each other, a fast roll-off filter has a lot more pre and post ringing than a slow roll-off filter. That pre-ringing is one of the reasons some people have complained about digital playback and kept using analog sources (Vinyl, tapes…)

So by choosing a fast roll-off over slow roll-off, many DAC and sound card makers are willing to sacrifice the phase and impulse response in order to have a nice looking RMAA graph.

However, while the slow roll-off filter minimizes the pre-ringing, it doesn’t get rid of it entirely. There is a growing number of CD players and DACs that provide a new option which the Minimum phase filter. According to their research, the human ear would be less sensitive to post ringing than to pre-ringing.

For those who are interested, Ayre wrote an interesting paper on Minimal phase filters (here: http://www.ayre.com/pdf/Ayre_MP_White_Paper.pdf)

Again those limitations (pre and post response) are far worse at 16/44 than they are at 24/96.
That is once again (in my opinion) the reason why some people hear a lot of improvement when using up-sampling or native 24/96 data. I think that despite some measurements (SNR, THD), not many systems have a true resolution greater than 16 bits. If that were the case (let’s say 20 bit resolution), those systems would be indistinguishable from the reality which is not the case. So, playing back 24/96 data on the DACs can improve the playback experience in comparison to 16/44 simply because most digital filters are terrible at handling 16/44. That is perhaps one of the reasons that we see so many DACs today that use ASRC/upsampling. (But then again those ASRC chips do not seem to sonically sound good and can also benefit from a good upsampling from the source, but that is another story …)


SoX upsampling in Foobar:

Among the nice things that come along with using a computer as a source is the ability to try different playback methods and upsampling methods.

Personally, I have settled for Foobar for quite a long time. And recently, I have been using SoX for 2 reasons: It is a relatively transparent upsampler (it doesn’t harm low level details) and it is very customizable.

In my case, I have found the most consistent and enjoyable results with upsampling to 24/96 (weirdly it sounding better than upsampling to 24/88) and with the minimum phase setting. It generally results in a more coherent soundstage that is pushed a little bit further back, and the sound is usually smoother. But those changes vary from one DAC to another, so it is impossible to make generalizations about the effects.

I have also tried different passband settings. The Stock setting is at 95%. When set at 99% the soundstage shrinks and everything gets tighter. When setting the passband at 90%, the soundstage becomes a little bit bigger and the sound and everything sounds a little bit softer and less defined.
Those different settings are a great tool for fine tuning the sound.

While I don’t use the SoX all the time with my reference DAC (which uses a slow roll off DF1704 digital filter), I found that it improved considerably the listening experience with the other DACs I have on hand.
post #853 of 1712
Good job, nice summary.

I thought all modern non-NOS dacs have built in oversampling so they can apply the filters further away from the audio range. You could in theory turn off the DAC's oversampling and apply the oversampling (or ASRC).
post #854 of 1712
Thread Starter 
Quote:
Originally Posted by glt View Post
Good job, nice summary.

I thought all modern non-NOS dacs have built in oversampling so they can apply the filters further away from the audio range. You could in theory turn off the DAC's oversampling and apply the oversampling (or ASRC).
You are right all non-NOS (by definition) apply oversampling. However, it seems that most built-in oversampling algorithms do a poor job at 44.1.
I guess that in order to do the oversampling (or upsampling) right you need a lot of raw power to do the math without too much damage. That is why many companies are investing in developping DSPs to "relieve" the built-in filters of the DAC chips. I hope my answer is not confusing.
post #855 of 1712
My external sound processor does have both optical and digital coaxial.
I'm using it for my CarPC audio.
This M2Tech sound very tempting. I'll definitely give it a try. Thanks for the respond.

Quote:
Originally Posted by slim.a View Post
The M2Tech hiface is a digital to digital (usb to spdif converter) while the X-Fi Surround 5.1 USB seems to be more a usb soundcard with after thought optical output. Do you use your X-Fi with an external DAC? If so, and if you DAC accepts coaxial, you will most probably get an improvement in playback performance (bit perfect, lower jitter, ...)
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Head-Fi.org › Forums › Equipment Forums › Dedicated Source Components › USB to SPDIF converters shoot-out : EMU 0404 USB vs. Musiland Monitor 01 USD vs. Teralink-x vs. M2Tech hiFace