I followed this guy for a long time until it was obvious that he quit and did not mean to return.
The design of that amp was frustrating. The power, the input and the output are all on the front of the amp. I always wondered why nobody cleaned that up and put power and input at the back. I also wished it used DC rather than 12 VAC.
Now I know, it was the no-derivative license.
When I was young I would mess around with audio circuits, trying different opamps, seeing if they changed the sound. These days I am far more concerned about connectivity.
His Objective DAC is more interesting in my opinion. If I ever need a 24bit USB DAC, I will get one. Considering I think 256 bit MP3s sound just fine, and FLAC far to much effort, I probably will never need one.
NwAvGuy proved what he wanted to do, that simple circuits are good enough and doing things like hand matching resistors with very expensive equipment is not required for a clean reproduction of audio. I still wonder if he really cared, or if he was a troll (he spend a good amount of time getting himself banned on forums).
Well, he got himself banned drom audiophile forums from pointing out their inconsistencies and ridiculous claims. I thought that was pretty cool of him, since he had the means to back it up.
edit: I forgot to agree with you on the design of the damn amp. I never got one either, because everything was in the damn front of the little box. When friends ask my opinion, I just tell 'em to get a schiit modi/magni combo. Looks way nicer, is a little cheaper, and you won't be able to tell the difference in the audio anyway.
To be fair, his style was highly argumentative. He left no room for subjectivity or talking about subjectivity in anything. Which leaves out a fair portion about what audio really is to the human brain, whether or not you perfect its reproduction mathematically.
His point of view, and his way of presenting it, were far too unbalanced and inhumane, regardless of their correctness. This was the reason he was not received well; not for the facts themselves.
This fact generalizes to a great many technical problems.
From what I've seen, NwAvGuy and other objective-audio types generally tend to have a better understanding of the subjective parts of audio though. It helps if you don't treat it as a mystery that can be affected by changes in the audio we can't measure. (Also, they're rather less likely to design amplifiers that destroy the equipment they're connected to than certain audiophile companies...)
> Which leaves out a fair portion about what audio really is to the human brain, whether or not you perfect its reproduction mathematically.
Hey I have a question, it's kinda off-topic but maybe someone reading this thread can answer this for me. Your comment made me think about this because I'm not sure if the effect I'm hearing is subjective, psycho-acoustic, physical or mathematical in nature.
I've been coding my own audio-synthesis toys on and off as a hobby for over a decade now, I like to believe I understand a thing or two about it :)
A little while back I wrote some very simple code to synthesize waveforms from the summation of sine-waves.
Summing sine-waves of frequencies 1..N with amplitudes 1/N produces a very nice bandlimited sawtooth waveform. This works, it sounds crisp, like a sawtooth, exactly what you'd expect.
Then I started thinking about phases. This is the important part: I have always understood that the human ear cannot perceive phase. I'm specifically talking about a mono signal (no L/R phase differences), perceived through a headphone (no interfering room acoustics), a continuous waveform (so any phase-shift only happens modulo 2pi maximum).
Obviously shifting the phases linearly with respect to the frequency just results in the same sawtooth wave shifted in time. Works exactly as expected, no audible differences.
So I continued with other ways of meddling with the phases, which resulted in radically different-looking waveforms. Adding a constant amount gives you a kind of comb-like impulse train with rounded bottoms. Setting the phase to frequency squared (times pi/3 or pi/5) gives you some really funky pointy looking shapes. Frequency cubed (again times pi / some integer constant) looks like a sum of a series of sloped square waves. Picking a (fixed) random phase for every component, gives something noisy-looking (but periodical, of course).
Pretty cool, really. FFT analysis of these waveforms confirms they still have exactly the same frequency components as a regular sawtooth wave. I triple checked, it works exactly as intended.
But here comes the twist: they sound different!
I checked on headphones, speakers, at varying levels of volume/amplitude to see if there was any non-linearities causing the difference in sound somewhere along the signal chain (nope, they sound different in exactly the same manner regardless of amplitude). If I had to describe he difference, I'd say they sound a bit more "hollow", in the sense that a square-wave sounds hollow, but not quite (and they obviously still have all their harmonics). It's not a subtle difference though, I can hear it very clearly.
What gives? I thought we couldn't hear phase, only phase differences, and phase cancellations? As in, only when one phase interferes with another, along the same frequency.
It kinda upsets the very foundations on which my whole mental model of audio DSP has been based: phase is irrelevant, unless either: A) phase shift is large enough to cause an audible delay, B) phase difference between left and right ear, C) two signals interfering with phases on the same frequencies, or D) you're doing some non-linear post processing. And I've never really seen anything written about it in literature, otherwise.
Anyone got an idea? How can a steady periodic signal sound different even if the frequency components amplitudes stay the same, but only the phases are shifted? And in what way does this affect the sound, I have a fairly good intuition of how frequency amplitudes change the sound (EQ, filtering), but not the phases.
My instinct is that the phase is causing some distortion somewhere along the audio path, between the signal (the ideal) and the analog. Between that is the DAC, the amplifier, the wiring, and the transducers, all of which can induce distortion of varying degrees based on various properties of the input.
But your experiment validates the simple concept that small, seemingly insignificant differences in audio can cause unexpected and perceptible effects! Awesome.
I read him differently. He only criticized objective claims which were scientifically implausible or found to be untrue, e.g., that vinyl had higher fidelity than CD as a medium, that humans could hear a difference between cheap and expensive cables, or when his measurements differed remarkably from manufacturers' specs, i.e., false advertising.
He did not criticize subjective preferences such as vinyl over CD, fancy cables over cheap ones or tube amps over op-amps. Only when false/implausible objective claims came into play did he refute them.
In case you're wondering, everything was at the front because he was trying to hit a price point for DIYers, and the custom-cut end panels were a big line item. By putting everything in one panel, the other could be left unaltered at a savings.
You can certainly argue that it was penny-wise pound-foolish, but that was the reasoning at least.
Also a more liberal license would have allowed things to be moved around, true, but it wouldn't have been an O2 anymore - the routing of the board is very important to its performance. I think perhaps a better choice on the licence would have allowed derivatives, so long as they were renamed significantly.
> If I ever need a 24bit USB DAC, I will get one. Considering I think 256 bit MP3s sound just fine, and FLAC far to much effort, I probably will never need one.
He decided for a 24-bit DAC in order to have enough headroom for software volume adjustment, because lowering the volume in software reduces the effective number of bits and thereby maximum dynamic range. He didn't claim that humans could hear a difference between 16-bit and 24-bit bit depth under practical conditions.
> His Objective DAC is more interesting in my opinion. If I ever need a 24bit USB DAC, I will get one.
FWIW I purchased his Objective DAC (from this site: http://www.headnhifi.com/odac-rca) and I don't hear less noise from this than my internal sound card (Terratec Aureon 7.1-Space).
So I was a bit disappointed. But perhaps my internal sound card was just that good, although I think the noise level is greater than is reported online. Audible noise at normal listening levels.
I have an ODAC and I can't detect any noise unless I max-out the volume knob with my ER-6is. Not that it matters, since playing anything at that volume with those headphones would be deafening and painful. I definitely hear more noise with my FiiO E7, and my laptop's headphone-out is even worse.
In the hope that I have some idea of what I'm talking about, I've listed some possible reasons why you may be hearing noise and how to fix them.
1. Using analog inputs (RCA or line in) instead of USB. This will just amplify the noise from your input source. Analog inputs are enabled even if you're playing sound through USB, so be sure to disconnect any secondary source from your amp.
2. Nearby EMI. I found I could hear a high-pitched whining if I put my amplifier near the lower-right of my monitor. Repositioning your amplifier and/or headphone cable may fix the problem.
3. Grounding issue (unlikely). Try unplugging the amp's adapter and using the DAC-line output[1]. If there's no noise, the problem must be in amplification. That means either a defective amplifier circuit or a grounding/EMI issue.
4. Bad software (very unlikely). I once had a copy of VLC that output low-level noise when paused. This was when x86-64 was new. There were plenty of other problems with VLC on 64-bit at the time, since it used ints and longs as bitfields. You'd have this problem with all soundcards, not just the ODAC.
It might also be useful to describe your set-up in more detail, including what headphones you've tried your amp with.
Edit: Another thing you'll want to do to improve audio output: Make sure your OS is sending 24-bit audio to the ODAC. On OS X, open up Audio Midi Setup, then select the ODAC output and switch it to 24 bit. The default is 16-bit, which reduces dynamic range if you don't have the software volume at 100%.
Make sure your sound drivers in your OS are set to use 24bit output only. I've had very noticeable noise when an audio stream was paused that went away completely when I changed the driver settings.
Any proper audio interface should have a very low noisefloor. It's more about stuff like phase distortion and crosstalk. You need to have a decent amp/speakers already to hear the difference.
Your noise problem will be coming from somewhere else. Probably a ground loop.
It's got a built-in battery charger and can also run from a wall wart without using batteries at all. The reason the external power supply has to be AC is that it needs dual negative and positive power supply rails, which it gets from rectifying the AC and using that to charge a pair of batteries. Cheap, easy, and in theory gives decent audio performance.
If you read NwAvGuy's blog, the modifications necessary to run it off DC are pretty much exactly what he wouldn't want people to do.
Beware of Newsweek if this story becomes too popular they'll start a thorough and extensive work of research and finally claim NwAvGuy is no other than Dorian S. Nakamoto.
This actually reminds me of ripster55[0], who is (in my mind) the leading expert of mechanical keyboards online, and is banned from many of the popular mechanical keyboard forums.
It's seemingly not an uncommon pattern in electronics related hobbies, you'll have a few main forums with their own orthodoxy, and a banned former member who disagrees, and clearly has at least a good idea what they're talking about.
For instance, rc multirotors has "timecop", who was banned from rcgroups. He designed a flight controller which is popular and well thought of. AIUI, he also did the hardware design for what are probably the most recommended enthusiast motor controllers today.
There are already companies out there producing modified versions of his designs which technically break the license. See the version from Head n Hifi http://www.headnhifi.com/O2-ODAC-fully-modded
I think if we were going to see him surface again he already would have.
Always wondered about audiophiles. Seems like a simple truism: The best quality reproduction of a recording will come from whatever the final mix engineer was using. If he was using a 400 dollar pair of Sennheisers and similar equipment, I'm not sure how you could possibly do better. Maybe you could do better if you had the masters and could remix it with more expensive crap, but that's unrealistic. And if the engineer really screwed up, no pricey amp is going to fix it.
Not exactly. Sound production and sound reproduction are separate pursuits.
Sound reproduction should ideally be input transformed with transducers to a listening environment. In the absence of perfect transducers there's a lot of play area for enthusiasts to modify the transformation stage to interplay with what is by far the weakest part of the chain, the transducers. Massy complex mechanical circuits with all the imperfections that entails. Despite this, it's very possible to improve the state of the art.
Sound production on the other hand is whatever the producer wants it to be. And the best transducers are often purposefully avoided for monitoring, because it won't necessarily sound too good on the radio when driving to work, or an MP3 through Dr Beats on a noisy tube or metro.
The recording might capture a lot more detail than what the sound engineer was hearing. For instance, the recording could have been made in the 60s or 70s, and the state of the art in sound reproduction can have improved since then. So unless you want to argue that hearing that extra detail that the mix engineer didn't hear is "wrong", I don't see how you can make your point.
If someone records a live concert using a pair of high quality mics and a pair of crap headphones, is the "truest" reproduction of that recording made with the crap headphones, or is the "truest" reproduction what sounds closest to being present at the concert?
I think we agree on the 24/192 thing being foolish, but the one significantly positive aspect of Pono is that it would allow access to remastered recordings without the loudness-wars style compression (Californication being a great example). My hope is that it will be a market that you can safely sell high-dynamic-range audio to because presumably they value that over preceived volume.
Eh, I'm not holding my breath on that (though of course I hope I'm wrong). In all the Pono marketing material I've read I've seen almost no reference to (re)mastering; it's all been about chasing the hi-res dragon. Meh.
I was speaking to him within a day of when he was last heard from. I got the feeling he was getting sick of the 'fame', I know he was getting a lot of flack from products he measured / reviewed from creators of high-priced, low quality manufacturers.
I'm not sure this is really that remarkable, he just had a personality and got some attention.
There have been high-quality amplifiers around for a long time now and Class T (http://www.google.com/patents/US5777512) amplifiers at that pricepoint are over a decade old. They're very popular with DIY audio guys. I've had my Trends TA-10 close to 10 years old now.
He did something cool, but I don't see it as all that remarkable.
I'm a pretty intensive electronics enthusiast. I checked out his site, and skimmed some of the materials.
My impression is that he deliberately designed a fairly mundane circuit using commonplace parts, to show that an esoteric design is unnecessary. So it's a "political" design, if you will.
The impressive thing is that his documentation of design decisions and techniques is pretty exhaustive. I've definitely bookmarked it for a closer read.
Off topic, but I found this layout painful to use, with something like 60% of the window devoted to links for other articles and the actual content squeezed into a column on the left. Why would anyone intentionally make things so hard to read?!
Couldn't we just "reverse engineer" the thing, and rebuild a design "from scratch"? I mean, his work should allow us to go back to first principles, from which we can rebuild a new design.
That's always a possibility, but I think most people who build his designs don't know how they work. High-quality audio reproduction isn't mysterious, but it's not trivial either. One must have a good sense of how and why amplifiers distort signals. That limits the possible candidates for a new design and assures that many will simply build a proven design they don't understand.
The design of that amp was frustrating. The power, the input and the output are all on the front of the amp. I always wondered why nobody cleaned that up and put power and input at the back. I also wished it used DC rather than 12 VAC.
Now I know, it was the no-derivative license.
When I was young I would mess around with audio circuits, trying different opamps, seeing if they changed the sound. These days I am far more concerned about connectivity.
His Objective DAC is more interesting in my opinion. If I ever need a 24bit USB DAC, I will get one. Considering I think 256 bit MP3s sound just fine, and FLAC far to much effort, I probably will never need one.
http://www.jdslabs.com/products/48/o2-odac-combo/
NwAvGuy proved what he wanted to do, that simple circuits are good enough and doing things like hand matching resistors with very expensive equipment is not required for a clean reproduction of audio. I still wonder if he really cared, or if he was a troll (he spend a good amount of time getting himself banned on forums).