O2 AMP + ODAC - Page 63

Quote:

Originally Posted by lorriman 


He does somewhere: by changing to 24bit mode.

well, i couldnt find it... ugh, nevermind, i guess ill have to take his/your word for it. perhaps ill get an answer one day... thank you though

How exactly does one change to 24bit mode in Windows 7?

Quote:

Originally Posted by cel4145 


Yep. Go into Control Panel->Hardware and Sound->Sound. Select "Properties" for the ODAC and go to Advanced.

I like it. 24-bit sound with the O2 and ODAC.

Quote:

Originally Posted by MrEleventy 


volume control in Windows control the playback device, not Windows output. the dac will get the full bit info and then scaled down from there. post conversion

well that explains it, thank you very much! so i guess digital volume control really is "guilt free" now, so long as i keep it reasonable.

Quote:

Originally Posted by lorriman 



It took ages but I finally tracked it down. This google link will take you to it "Why 24 bits can matter" :


I'll just add my ha'penny to clarify the O2 volume control:aAs I've understood it, the O2 works rather differently to many amps (he says not having actually any clue about amps or electronics). Instead of lowering the volume before the audio is pumped in to the amplifying electronics, the voltage is lowered AFTER the amp circuit using a 'voltage divider'. I've no idea what all that means, really, nor the implications as I'm not a genuine electronics person. Perhaps someone can tell us.

lol, you may want to edit that link out, it may not be a direct link, but its still kinda cheeky i think 

anyhow, i already read all that, what i was trying to figure out is what MrEleventy was kind enough to explain to me. and id like to point out again that in my personal experience, i personally couldnt tell a difference between 24 bit, and 12 bit depth. i could only notice the differences at around 8-10 bits, and even then, it was mostly just background noise. ill recommend again that you search youtube for ethan winers link where he demonstrates this with a program that "strips" away one bit at a time. i think he even provides a download link so that you can try it for yourself.

not an electronics guy either, so i cant help you there im afriad. thank you for your comments, much obliged =]

Just put my brand new 1 week old JDS stand alone O2 and ODAC up for sale on the forums if anyone is interested :D

Hopefully nobody yells at me for responding to old posts.

Quote:

Originally Posted by MrEleventy 

 

My current setup is E17 LO > O2 > M80s, looking at the specs, it's not too far off from the M-100s (28.5 ohms vs 32 ohms, 105db vs 103db), what I do, since all 99% of my files are 16 bit anyways, I set my E17 to run at 24bit and lower the volume by about 25% and that gives me more wiggle room on the O2. I don't know the exact math but doing my own fuzzy math, I figure since 16 to 24 is about 66.7%, I can throw away 33% before losing "quality". I actually don't notice a difference at 50% but my OCD goes off and I crank it back up. 

 

P.S. My O2 is still on the defaulted 2.5x/6.5x, FWIW.

16 bits to 24 bits is a difference of having 2^16 = 65536 possible values vs. 2^24 = 16777216 possible values, or a factor of 2^8 = 256 difference.  So around 20 * log10(256) = 48 dB.  Theoretically you could throw away 99.6% of the signal.  Note that 66% in some software volume control setting may not actually correspond to 66% of the signal.  The scaling often doesn't work that way.

If you're playing a 16-bit file with 24-bit output at 100% volume control, let's say those 24 bits for a particular sample are the ones below, as a simplistic example, ignoring a couple small details.  (The order is from most to least important starting from left.)  Notice the rightmost eight values are 0 because you only have 16 bits of information.:

01010110 00100111 0000000

If you reduce the volume some digitally, the output will be essentially shifted to the right:

00010101 10001001 1100000

With a 16-bit output device, the two things would look like

01010110 00100111

and

00010101 10001001  (whoops where'd the last couple bits go?)

In practice, no 24-bit output device can achieve 24-bit resolution because internal noise (the output and process is analog, after all) is significantly higher than any amplitude change attempted by any shift in the last few bits.  ODAC effectively "only" has 19.3 bits of resolution because of this, so don't expect to reduce volume by 40 dB in software and not be effectively losing resolution under the noise floor.  Many other 24-bit output devices are worse than that, but some but definitely not all more expensive ones are better. 

Note that in practice, most recordings have noise from the recording studio / mics / etc. that are above the 16 bits, and 16 bits may be overkill also depending on listening volume, room ambient noise level, etc.  As mentioned by somebody, check the YouTube video and link of files for Ethan Winer's audio myths workshop.  There's a demo there of reduced resolution.  Mostly you just get a bit of noise, which may not be noticeable until well past 16 bits, depending.

Quote:

Originally Posted by adamlr 

 

thank you, but im not worried about wearing them out, im worried that with time, constantly being plugged in and charging, theyll die and damage the amp, not necessarily the most reasonable fear, but id still rather just take them out, leave the amp plugged in, and forget about it.

 

thank you, so i was half right? *1 does infact mean that your left with the same voltage, i guess thats why its called "unity gain"? so, *1 gain means that you only add current? and this still sounds the same? doesnt it react differently to your headphones specs as opposed to "regular" gain?

1x gain is not so much adding current as adding the ability to deliver more current if necessary.  The current actually being delivered depends on the output volume level and the load impedance.  The electronics and structure are different for one amp as compared to another—there may be a difference in noise levels, output impedance, frequency response, nonlinear distortion at different loads / output levels / frequencies and so on.  This may or may not translate into some reliably-perceptible difference in sound quality.

Actually, with 1x gain, you are amplifying any noise from the input by a lesser amount, so that is slightly better, possibly, along with more negative feedback for the gain-stage op amp (which is not the limiting factor; the output stage op amp is).  Probably not any difference in practice.

Quote:

Originally Posted by lorriman 


There's a further complication in that there may be a difference between XP and Vista/7. the suggestion I'e seen is that XP is affected by the 16bit volume issue (necessitating max source volume) but Vista/7 aren't. Apparently XP had an infamous sound system that would do this bit-stripping, as I've seen it called, but Vista/7 aren't affected. It would be good to get a clarification of that. Perhaps it's because Vista/7 default to 24 bit, or perhaps they talk to the amp circuit directly somehow. However, the ODAC horrifyingly defaults to 16bit on my Vista computer which I didn't know for some time, so perhaps not the former. Of course, this could be entirely bogus: in that all 3 OS's are affected equally.

In any case, I tried XP at 16bit mode and reduced source volume to a minimum using my O2 to amplify: I didn't hear much difference. But I would have to repeat that test to be sure of my claim there which I can't do at the moment.

The main things is that most modern sound cards can be easily switched to 24bit mode on Windows which means you can turn the source way down, with all the advantages that brings (and one disadvantage). I would be surprised if there were not some way of verifying that on your MAC, or changing it. (The disadvantage is that some forms of noise can be made inaudible by turning the volume to max on the source and turned down on the amp/O2. )

There really shouldn't be a difference between OSes in how the volume is handled.  Barring some kind of digitally-controlled analog volume control or some trickeration on the DAC side, volume should be reduced digitally by shifting the bits over so the effective output value is smaller, thus squashing the range down (and throwing away the least-significant bits, more or less).

Quote:

Originally Posted by lorriman 

I'll just add my ha'penny to clarify the O2 volume control:aAs I've understood it, the O2 works rather differently to many amps (he says not having actually any clue about amps or electronics). Instead of lowering the volume before the audio is pumped in to the amplifying electronics, the voltage is lowered AFTER the amp circuit using a 'voltage divider'. I've no idea what all that means, really, nor the implications as I'm not a genuine electronics person. Perhaps someone can tell us.

Right, many audiophile amplifiers use a potentiometer (variable resistor), which is controlled by that volume control knob, to divide down the signal level prior to any gain being applied.  A voltage divider is just a fancy name for usually a couple of resistors in series; the total voltage across the combination of the two is divided between the two resistors—e.g. if one has 100 ohms and the other has 200 ohms, the one with 100 ohms gets 100 / (100 + 200) = 1/3 of the signal, while the one with 200 ohms gets 200 / (100 + 200) = 2/3 of the signal.  O2 potentiometer is after the gain, which is apparently more of a pro-audio style configuration.  It leads to problems of the gain stage clipping if the input level and gain are too high, because the signal level essentially becomes too high for the gain stage to handle.  If the volume control attenuates the input, then you can set the level to be low enough such that it's not clipping the gain stage.

Depending on the main contributors of noise in the system, doing the volume control after the gain could improve the SNR.  More or less it's keeping the signal level high so it is higher relative to the noise level.  The volume control divides down both the signal and the noise (mostly).  Suppose for example that the gain stage contributes some X amount of noise on its output.  If you reduce the signal level prior to the gain stage, the output signal of the gain stage will be lower relative to X, so you'll get a lower SNR.  If you keep the volume control after the gain, the output of the gain stage is large compared to X, and both the output and the noise get reduced by the volume control (somewhat), so the higher SNR is maintained.

Some people say the O2 has way overkill low noise levels, and it would be better to just have the volume control like how most other audiophile amps are, so people wouldn't have that clipping issue with a dumb setting.

**** yeah I just received my o2 and odac  SigDJ + odac + O2!

Does anyone here have experience with the Leckerton UHA-6S? I'm curious as to how it compares to the O2 as a more portable option.

Quote:

Originally Posted by miceblue 

Does anyone here have experience with the Leckerton UHA-6S? I'm curious as to how it compares to the O2 as a more portable option.

Not me, but I seem to recall that some of the pirate crew (purrin, Anax, maybe others) had heard both and preferred the Leckerton.  Of course, these are the guys who think NwAv's getting money under the table for the O2 / ODAC.

Seems like Mr. Leckerton quite knows what's up, is an application engineer at Cirrus.  UHA-6S MKII uses a high-end Cirrus Logic DAC.  There are some measurements of the older version published on the site, not completely comprehensive, but let's assume the rest checks out too.  According to the objectivist camp and so on, there probably shouldn't be a discernible difference in sound between the two devices, operating at levels before clipping.

So at least by the specs, the limitations compared to O2 / ODAC are:

• limited to 16/48 over USB (but there is an S/PDIF input)
• lower output power levels — 30 mW @ 16 ohms, 55 mW @ 32 ohms, 110 mW @ 100 ohms, 55 mW @ 300 ohms (presumably with stock output op amp)

  O2 gets 353 mW @ 15 ohms, 534 mW @ 33 ohms, (interpolated by me) 272 mW or so @ 100 ohms, 94 mW or so @ 300 ohms, on a mid-high charge on battery

So the O2 is better for some planar magnetics, maybe AKG Q701 if you listen really loud.  Otherwise, no lower-impedance sets actually need that kind of power.  O2 is also a little louder for high-impedance sets.

Supposedly the older UHA-6S has higher noise than the MKII, and even though there's no noise level listed, you can look at the spectrum on a THD+N spectrum graph for the original UHA-6S and see most of it around -145 dBV or so, whereas the O2 is around 5 dB quieter across the band.  So at least the amp portion is really really quiet, good if you use some super-sensitive IEMs.

Listed battery life is better than that of the standard O2.