[stv014]

Quote:

This is a myth. Software volume control is not harmless. It's theoretically correct for an ideal DAC with perfect 24 bit resolution, but for a practical device, you have to work from 0dB and consider what the device's actual resolution is, because digital attenuation throws away the high order bits first, not the low order bits.

 
That is nonsense. If digital attenuation would throw away the highest order bits first, the result would sound extremely distorted and almost like white noise. Additionally, properly implemented attenuation dithers the final 24-bit integer result, and does not just throw bits away.
 
Quote:

Originally Posted by MoonUnit /img/forum/go_quote.gif
 
At full scale, the ODAC has 0.0029% THD+N, which means the noise and distortion floor is at -90.7dB. It's irrelevant that the dynamic range is higher than this, unless one believes that dynamic range is more important than distortion. When you digitally attenuate by 40dB (fairly common), you're producing a device whose noise and distortion floor is now at only -50.7dB. This is in the realm of low quality tube amp territory, and will be definitely audible.

 
You are assuming that the absolute level of distortion remains constant, but that is false. Check the dynamic range measurement graph, with the -60 dBFS input, the distortion products drop to not much higher than -130 dBFS. Additionally, if you digitally attenuate by 40 dB (which is a rather high amount, by the way), you also make the sound overall much quieter, and thus the distortion/noise at the same THD+N will be more difficult to hear. If the absolute noise level is below the hearing threshold at full volume, then reducing the digital volume will not make it audible, even if the dynamic range is reduced, because at low volume the dynamic range of hearing is reduced, too.
In fact, many DACs have increased distortion with signal levels near 0 dBFS, or clip peaks that would go above 0 dBFS after the digital filter. Therefore, a few dB of software attenuation can even reduce distortion.
 
Quote:

Originally Posted by MoonUnit /img/forum/go_quote.gif
 
You can even see this on the graphs on NWAVGuy's site. I realize NWAVGuy makes his recommendation because he believes distortion below 0.1% (-60dB) is inaudible, but he has never given any support for that assertion and it is not consistent with the academic evidence from listener tests, by a wide margin. See, e.g., this peer-reviewed publication: http://www.aes.org/e-lib/browse.cfm?elib=2962  (0.003% is -90.4dB)

 
He does not claim that distortion below 0.1% is inaudible, the recommended maximum is actually 0.01%. It is also difficult to comment on your peer reviewed publication - which refers specifically to one type of distortion, high frequency IMD, and may not even be relevant to the topic of digital volume control - without having access to its contents, and knowing the details would be important. Many old publications have also been proven wrong later, after unexpected flaws in the methodology were discovered (e.g. distorting tweeters making ultrasound "audible", etc.). Fishing in old papers, one could find something to support just about any agenda.
 
But if you give a link to some music of your choice, I can create for you a recording that goes through -20 dB digital attenuation and a D/A-A/D loop, and you can compare that to the original in a level matched software ABX test with whatever gear you have, and see if you can tell the difference at a realistic loudness (not more than 90 dB peak SPL because the sound is supposed to be attenuated, after all, so full volume would be a deafening 110 dB).

 

[MoonUnit]

Quote:

I'm not suggesting using digital attenuation is completely harmless, but I am suggesting that in most situations it shouldn't make any substantial, audible difference. Especially when using a properly implemented 24-bit DAC.

 
I would encourage you to test that assumption, as it is not consistent with the academic literature on distortion audibility.

 

[MoonUnit]

Quote:

 
That is nonsense. If digital attenuation would throw away the highest order bits first, the result would sound extremely distorted and almost like white noise. Additionally, properly implemented attenuation dithers the final 24-bit integer result, and does not just throw bits away.
 
 
He does not claim that distortion below 0.1% is inaudible, the recommended maximum is actually 0.01%. It is also difficult to comment on your peer reviewed publication - which refers specifically to one type of distortion, high frequency IMD, and may not even be relevant to the topic of digital volume control - without having access to its contents, and knowing the details would be important. Many old publications have also been proven wrong later, after unexpected flaws in the methodology were discovered (e.g. distorting tweeters making ultrasound "audible", etc.). Fishing in old papers, one could find something to support just about any agenda.
 
But if you give a link to some music of your choice, I can create for you a recording that goes through -20 dB digital attenuation and a D/A-A/D loop, and you can compare that to the original in a level matched software ABX test with whatever gear you have, and see if you can tell the difference at a realistic loudness (not more than 90 dB peak SPL because the sound is supposed to be attenuated, after all, so full volume would be a deafening 110 dB).

 
It is not "nonsense". My reference to "throwing away" bits was by way of analogy to explain the problem to people with a computer background, as it's usually computer people rather than engineers who make this error. Obviously I am assuming a properly implemented 24 bit volume control with dither. It doesn't change the result at all.
 
I noticed you edited your post to remove additional details that you were incorrect about, so I will not comment on them. Suffice to say that if anyone doubts that digital attenuation raises the distortion and noise floor essentially in a linear manner, look at the THD+N vs. level graph in the datasheet of your converter. For example, see here:
http://www.ti.com/lit/ds/symlink/pcm1704.pdf
That's a much higher performance 24-bit DAC in terms of distortion (not necessarily other parameters) than the chip in the ODAC. Look at the first graph on page 4. You can see that they hit 0.01% distortion (the threshold you're advocating) using 24 bit data at just -26 dB of digital attenuation.
 
Your suggested test is valid, but you have to use realistic levels of digital attenuation, because most people do not have the gain structure of their system properly calibrated. The default gain of the O2 combined with the nominal output level of the ODAC, for example, is going to mean that many people will need around 40-50dB of attenuation for typical headphones, particularly low impedance headphones. I have a very carefully calibrated gain structure in my own speaker system, and I still need 16dB of attenuation at a minimum. Use realistic numbers and you may be surprised by the result. Also, you are confused about what listening level the test must be run at. The whole point of attenuation is to adjust the gain structure to reflect a person's standard listening level, so you must run the test at a person's standard listening level.
 
In any case, regardless of whether the additional (substantial) distortion is audible, there's no reason to introduce it at all. In circumstances where analog attenuation is an option, it doesn't make engineering sense to introduce 20-50 dB or more of additional distortion *relative to your standard listening level* by using a digital volume control. Why buy the ODAC at all in this case?

 

[skamp]

The default gain of the O2 combined with the nominal output level of the ODAC, for example, is going to mean that many people will need around 40-50dB of attenuation for typical headphones, particularly low impedance headphones.

Uh? I'm fine with 1x gain, no digital attenuation and the volume knob at 10 O'clock, with my Denon AH-D2000 (25Ω / 100-106 dB/mW).

 

[mnarwold]

I received my ODAC and O2 today.
Without any burn-in (Does this benefit from burn-in?) it sounds pretty good. I'm using J.River to output 24bit using WASAPI -Event Style (something the HRT Headstreamer had problems with). Keeping the system volume at 100% I don't even go above 9 o'clock on my Grado SR225i. Of course, those are pretty easy to drive in the first place.
Unfortunately, they forgot to ship the power source, so I'm running it off a pair of 9v batteries. Is the SQ the same with the batteries and the power supply or does the power supply sound better?
I can't compare this directly to my Headstreamer since I sold it before I received my new setup. I tested the Headstreamer against my iPod 5.5 and then compared my iPod to the ODAC+O2 and think the ODAC+O2 is a little more revealing. In Cry Me A River by Michael Buble, the background instruments had more nuance. Rather than a particular not being played, I could tell that it was a saxophone and heard a little bit of the buzz/blare that is normal for it to produce. I couldn't tell a difference on the foreground instruments, just in the background.
 

 

[adydula]

Hey glad you got your ODAC and O2!
 
I run mine mostly from batteries, and attach the ac adapter evey other day overnight to re-charge them.
 
I honestly do not hear any audible differences running with the ac adapter attached vs running on pure "DC" from the batteries.
 
You would think that with all the discussions about power supplies etc being crap and having to use higher dollar sans higher quality supplies that there would be ......but by running on batteries all those AC power supply issues, filtering, transformer hum and noise, ac ripple etc all go away.....thats whats really neat about the O2 design....
 
I even bought several AC adapters at different voltages to see if this would improve the sound quality etc...to me nothing....they all sound just fine....12v to 18 volt.
 
With my Grado 325is which are what I call a bright set of cans..I noticed that these are even "brighter'' than before...especially with brass instruments....with vocals they are wonderful...
 
I have a set of LCD2's, the Grados and a set of AKG 701's and they all sound great with this combo....
 
Hope you get your ac adapter soon!! Before the batteries run down!!
 
lol
 
Alex

 

[mnarwold]

Quote:

Hey glad you got your ODAC and O2!
 
I run mine mostly from batteries, and attach the ac adapter evey other day overnight to re-charge them.
 
I honestly do not hear any audible differences running with the ac adapter attached vs running on pure "DC" from the batteries.
 
You would think that with all the discussions about power supplies etc being crap and having to use higher dollar sans higher quality supplies that there would be ......but by running on batteries all those AC power supply issues, filtering, transformer hum and noise, ac ripple etc all go away.....thats whats really neat about the O2 design....
 
I even bought several AC adapters at different voltages to see if this would improve the sound quality etc...to me nothing....they all sound just fine....12v to 18 volt.
 
With my Grado 325is which are what I call a bright set of cans..I noticed that these are even "brighter'' than before...especially with brass instruments....with vocals they are wonderful...
 
I have a set of LCD2's, the Grados and a set of AKG 701's and they all sound great with this combo....
 
Hope you get your ac adapter soon!! Before the batteries run down!!
 
lol
 
Alex

 
So I think my batteries ran out

  It just started popping randomly. I'm hoping it is just batteries and not something more serious.

 

[mikeaj]

Quote:

 
So I think my batteries ran out

  It just started popping randomly. I'm hoping it is just batteries and not something more serious.

 
Popping is what you can get when the batteries drain.  Actually, the nature and behavior of the popping depends on the actual batteries and certain resistor values in the circuit.  It's supposed to be shutting the amp off to prevent the batteries from being overdrained, that's all.  Sometimes, the behavior is a little funky and it keeps tripping on and off until you actually recharge the batteries again. 
 
Pop in some new batteries or wait for the adapter.  It should be good to go then.

 

[adydula]

Go get some cheap Duracell etc batteries....do you know how to take it apart?
 
Alex

 

[mnarwold]

I put in some cheap batteries already. I bought from Mayflower Electronics that gives you the option to not buy the rechargeable batteries for a discount. Tyler told me he would be sending me a gift to make up for forgetting the power supply and I'm kinda hoping it's the batteries. I don't really expect to be using my amp as a portable amp which is why I decided not to get the batteries. Of course, now I'm seeing where they'd come in handy

 
After letting the amp sit for a little while it is now working again

but who knows for how much longer
 

 

[Ralf Hutter]

What happens if you try to play music through the ODAC that is not at one of its' supported resolutions? I have many 24/88 SACD rips, and according to the data, that is not supported by the ODAC. Will these play? Will they be downsampled or modified in some way prior to being played?

This question was asked several times earlier in this same thread and it was never answered, as far as I can tell.

Thanks!

 

[adydula]

ralf...
 
You can find out more about this on the designers site...or google "sacd odac" and then pick the first topic....points to his blog, do a control F and search for SACD.
 
The nut of it is that some have asked for 24/88 high res support like yours from SACD rips. The ODAC doesn not support these. The workaround is to re-sample the 24/88 rips to the audibly identical 24/44 format. Its easy to do, there are not artifacts....a simple divide by two process. Some people have done comparisons with 24/88 to the 24/44 with Foobar and the ABX addon. Unless you mess up the re-sample you will not be able to tell them apart.
 
I dont have any SACD rips and dont know off hand what would happen if you tried to play them, most likely no sound etc....try it.
 
Alex

 

[rs01]

Quote:

What happens if you try to play music through the ODAC that is not at one of its' supported resolutions? I have many 24/88 SACD rips, and according to the data, that is not supported by the ODAC. Will these play? Will they be downsampled or modified in some way prior to being played?
This question was asked several times earlier in this same thread and it was never answered, as far as I can tell.
Thanks!

Ralf,
 
I can play 24/88 files I purchased off hdtracks without any issue (I just specifically tested https://www.hdtracks.com/index.php?file=catalogdetail&valbum_code=HD00028947419921) with Foobar and the ODAC.  FWIW I don't know if or how the downsampling would work, but it still sounds excellent.
 
rs

 

[Funky-kun]

Using foobar2000, when trying to output 88.2 khz files through WASAPI you get:
 
 

Unrecoverable playback error: Unsupported stream format: 88200 Hz / 24-bit / 2 channels

 
Under Direct Sound it plays fine, the OS taking care of the resampling, as set in the control panel. 
 
I personally avoid the issue by resampling only the unsupported sample rates with SoX to 44.1.

 

[Dano91]

Quote:

What happens if you try to play music through the ODAC that is not at one of its' supported resolutions? I have many 24/88 SACD rips, and according to the data, that is not supported by the ODAC. Will these play? Will they be downsampled or modified in some way prior to being played?
This question was asked several times earlier in this same thread and it was never answered, as far as I can tell.
Thanks!

It depends.
1.) If using WASAPI mode in foobar, then it will not play. But there is a workaround - just use "Resampler" in DSP Manager and choose any supported sample rate by ODAC (44.1/48/96)
 
2.) If using DirectSound, which is default in foobar, there will be no problem in playing 24/88, because Windows will resample automatically. (to sample rate chosen in Control panel -> Sound -> (ODAC) Properties -> Advanced)