[francisdemarte]

Quote:

The O2's enclosure is the same type as the recommended one for the AMB gamma1 (config F). The gamma2 also.
 

That how I was going to use mine. Stack it on top of the AMB Gamma2 for a nice transportable unit. I wonder if you can squeeze a GrubDac or BantamDac into the case too?
 
Check out http://www.diyforums.org/ for really small DIY DAC's

 

[JamesMcProgger]

 
Say if you put your cellphone on top of the unit while using the amp.
does the O2 caught cellphone interferences? with the original case open at the sides or the full enclosed one?
 

 

[maverickronin]

Quote:

Say if you put your cellphone on top of the unit while using the amp.
does the O2 caught cellphone interferences? with the original case open at the sides or the full enclosed one

 
There's almost none with my O2 and GSM cell phone.  You can hear a bit if they're right on top of each other but its not bothersome even with sensitive headphones.
 

 

[4nradio]

I finished the first of two O2 amps today. All of the initial tests went well and when I was finally able to plug in some "non-throwaway" headphones and listen to some FLAC files I came away impressed. I find the O2's sound to be very transparent and easy to listen to. I've tried some IEMs and Beyer DT-770 80 ohms headphones, but haven't yet plugged in my LCD-2 Rev. 2 headphones (need to get an 1/8" to 1/4" adapter).

BTW, the front panel from FPE is a perfect fit and alignment of all the machined holes is very accurate. They do great work...I look forward to seeing their facility in January at a Seattle Head-Fi enthusiasts meet.

An image of my first O2 amp is here:
 

 

[turokrocks]

Quote:

I finished the first of two O2 amps today. All of the initial tests went well and when I was finally able to plug in some "non-throwaway" headphones and listen to some FLAC files I came away impressed. I find the O2's sound to be very transparent and easy to listen to. I've tried some IEMs and Beyer DT-770 80 ohms headphones, but haven't yet plugged in my LCD-2 Rev. 2 headphones (need to get an 1/8" to 1/4" adapter).

BTW, the front panel from FPE is a perfect fit and alignment of all the machined holes is very accurate. They do great work...I look forward to seeing their facility in January at a Seattle Head-Fi enthusiasts meet.

An image of my first O2 amp is here:
 

Looks clean and nice, I ordered mine from 

Epiphany Acoustics

Hope it will be to my expectation, your (initial) review makes me impatient to get mine!!!
Please update us.
 

 

[shadow419]

I've been rocking my o2 for about two weeks.  Still waiting on my front panel from a group buy, but it sounds and performs like it should.  I'm using a a pair of HD 25-1 II.  I'd heard that they were a bit sibilant before from a lot of reviews, but I never heard it until I started using the o2 to power my cans.  I still love my HD 25's, but I can definitely hear them as they really are now.

 

[hans030390]

What would be a good DAC for this? If I get one of these amps, I'll be using my PC as my source. I'm currently using an Auzentech Bravura sound card. Would that be sufficient as a DAC? I would not be opposed to getting something different if it would work better. If the Bravura would be fine, are there any particular settings/configurations that work best with the amp (16-bits vs 24-bits, sampling rate, etc.)?

 

[Willakan]

Quote:

What would be a good DAC for this? If I get one of these amps, I'll be using my PC as my source. I'm currently using an Auzentech Bravura sound card. Would that be sufficient as a DAC? I would not be opposed to getting something different if it would work better. If the Bravura would be fine, are there any particular settings/configurations that work best with the amp (16-bits vs 24-bits, sampling rate, etc.)?

 
I can't speak for the Auzentech, but sample rate and the like should simply be set to automatically match the source material.
 
 

 

[hans030390]

Quote:

I can't speak for the Auzentech, but sample rate and the like should simply be set to automatically match the source material.

OK. It's just that I was reading something about the usage of (certain?) USB DACs (not sure if it applies to sound cards) set to 16-bit that would drop bits depending on the volume. I'm not very knowledgeable on some of these more technical aspects, so I don't remember exactly what I read.
 

 

[mikeaj]

Quote:

OK. It's just that I was reading something about the usage of (certain?) USB DACs (not sure if it applies to sound cards) set to 16-bit that would drop bits depending on the volume. I'm not very knowledgeable on some of these more technical aspects, so I don't remember exactly what I read.

It's not just USB DACs.  If you have software volume control lowering the volume below 100%, that's done by (more or less) dividing all the amplitude values by a certain amount.  Thus what the DAC gets is not the original, but a scaled-down version.  Because of the way the data is represented, when you scale things down, you effectively are rounding data out and losing it.  e.g. 186 -> 18.6 gets rounded to 19, so some info is lost (but it's not in base 10 of course)
 
If you have 16 bit output and using software volume control set fairly low, the data that is lost was potentially audible.
 
 
So set 24 bits (even if your music is 16 bits) since the hardware supports that, and then set the sample rate to the sample rate of the music.  That's 44.1 kHz for CD Audio, 48 kHz for DVD audio, and so on.

 

[Willakan]

You do get lower resolution using the software volume control - so don't use it. Best practise is to keep everything at full volume (no processing) in the digital domain and then use the analog volume control on your amplifier to adjust the volume level.

 

[mikeaj]

I did a quick assessment of the amp using RMAA (for what little that's worth) and my cheapo USB X-Fi Surround 5.1 sound card.  Measurements are limited by the line in on the sound card in many cases.  This is with default Windows drivers, so 24-bit modes don't work; I tested 16-bit 48 kHz only.  For some reason, the line in level is not calibrated right using default Windows drivers somehow, so the noise floor is a bit high I think. 
 
First test was sound card loopback (no load).  Then I tested with three different headphones in turn:  AKG K601 (120 ohms nominal), Beyerdynamic DT770 Pro (80 ohms nominal), Beyerdynamic DT 235 (32 ohms nominal).  After that, I tested with the sound card into the O2 (no load), and then into the O2 with those same three headphones.
 
My O2 has gains of 1x and 5.1x.  Below tests are for 1x gain, on AC power.  I started to test the 5.1x gain setting, but the results were all the same, so I got bored and didn't finish that.  I also spot checked with battery power only, but that looked the same too.  THD = total harmonic distortion; IMD = intermodulation distortion (SMPTE test).  Though really, these are looking at THD + noise, and IMD + noise.
 
All tests were at the max line out voltage of 1.1V rms, measured with a multimeter.  Actually, the sound card's output impedance is nontrivial, so the headphones were getting a little less than that on some tests, but not with the O2.  Click on graphs for large versions as usual.
 
Sound card (no O2) THD:

0.0027%, 0.018%, 0.025%, 0.046% (top to bottom, meaning unloaded, K601, DT770 Pro, DT235).
 
Sound card (no O2) IMD:

0.0098%, 0.025%, 0.096%, 0.050% (top to bottom).
 
 
Sound card -> O2 THD:

0.0028%, 0.0026%, 0.0025%, 0.0025% (top to bottom).
 
Sound card -> O2 IMD:

0.010%, 0.010%, 0.010%, 0.010% (top to bottom).
 
 
Again, note that results are limited by the sound card.  Also this is just RMAA.  Don't take too seriously.

 

[hans030390]

Quote:

It's not just USB DACs.  If you have software volume control lowering the volume below 100%, that's done by (more or less) dividing all the amplitude values by a certain amount.  Thus what the DAC gets is not the original, but a scaled-down version.  Because of the way the data is represented, when you scale things down, you effectively are rounding data out and losing it.  e.g. 186 -> 18.6 gets rounded to 19, so some info is lost (but it's not in base 10 of course)
 
If you have 16 bit output and using software volume control set fairly low, the data that is lost was potentially audible.
 
 
So set 24 bits (even if your music is 16 bits) since the hardware supports that, and then set the sample rate to the sample rate of the music.  That's 44.1 kHz for CD Audio, 48 kHz for DVD audio, and so on.

 
Gotcha, thanks! I'm assuming I'll be using the same line-out that I use to connect the sound card to my stereo receiver, then? Oh, and to clarify, does software volume control only refer to the volume control in software such as Foobar, iTunes, etc., or does it include the volume control in Windows as well?
 
I'm mostly concerned about my sound card (Auzentech Bravura) being a limiting factor to the amp...would that be the case, or should it be a good DAC for the amp?
 

 

[mikeaj]

Quote:

 
Gotcha, thanks! I'm assuming I'll be using the same line-out that I use to connect the sound card to my stereo receiver, then? Oh, and to clarify, does software volume control only refer to the volume control in software such as Foobar, iTunes, etc., or does it include the volume control in Windows as well?
 
I'm mostly concerned about my sound card (Auzentech Bravura) being a limiting factor to the amp...would that be the case, or should it be a good DAC for the amp?

Software volume control meaning the OS (Windows) master volume as well as those of any program.  This includes ReplayGain, any volume altering say for an EQ setting, and so on.
 
Auzentech Bravura might be a limiting factor to the amp in some sense, but I don't think it will be bad.  I think it's more likely that your music's recording/mastering, your headphones, or your ears will be the limiting factor first though, but that's just a guess.

 

[hans030390]

Quote:

Software volume control meaning the OS (Windows) master volume as well as those of any program.  This includes ReplayGain, any volume altering say for an EQ setting, and so on.
 
Auzentech Bravura might be a limiting factor to the amp in some sense, but I don't think it will be bad.  I think it's more likely that your music's recording/mastering, your headphones, or your ears will be the limiting factor first though, but that's just a guess.

Well, would I be able to find a better DAC under, say, $100-150?