O2 AMP + ODAC
Aug 18, 2014 at 6:44 PM Post #3,361 of 5,671
  You do not need 6x at all for Q701's...I had them for awhile etch... works well with unity gain or 2.5x.
 
Alex

 
Hmm, how loud were the headphones at default gain? I have the k712s which people claim to have nearly identical drivers, and at 2x the sound is quieter/fainter with distorted bass in my experience.
 
Aug 19, 2014 at 5:51 AM Post #3,362 of 5,671
I have gains of 1x/3x and that's more than enough for my Q's.
 
Aug 19, 2014 at 8:12 AM Post #3,363 of 5,671
Hmmm.... never heard distorted bass at unity or 2.5x if played at levels that don't destroy your hearing.
 
There are so many people that use the 02 amp at 2.5 levels and drive the Q701 and Q702 and Audeze LCD2's to ear splitting levels.....
 
If you listen to levels cranked past 12 o'clock etch you might be damaging your hearing over time.
 
Also the quality of the recording your playing has.a profound influence on its sound quality.
 
If your really hearing distortion with 2.5x gain setting then something is wrong with your setup or source.
 
A.
 
Aug 19, 2014 at 11:30 AM Post #3,364 of 5,671
My audio is set at 15/100 on the PC, 3 o'clock on the volume dial at 1x gain with the Q701's paired with ODAC/O2.
Once in a while I double the windows volume to 30/100 if I want to have "fun"
 
Aug 19, 2014 at 1:11 PM Post #3,366 of 5,671
I have my JRiver max volume set to 98%,a 2dB cut in effect. When it's set to 100% it will occasionally clip.
 
Aug 19, 2014 at 5:45 PM Post #3,369 of 5,671
I clearly advise people who don't have an impossible to drive 600ohm low sensi headphone, to get the 1x 2.5x version. but anyway the difference is a pair of resistors, so it's not really an important choice and it can be reversed easily enough.
 
 
Quote:
 
  You do not need 6x at all for Q701's...I had them for awhile etch... works well with unity gain or 2.5x.
 
Alex

 
Hmm, how loud were the headphones at default gain? I have the k712s which people claim to have nearly identical drivers, and at 2x the sound is quieter/fainter with distorted bass in my experience.

you simply never can make assumptions between gains. in effect you're just changing the resistor in the op amp and the resistor of the volume knob at the same time to get pretty much the exact same thing. mostly the difference will be clipping level and you most likely will only ever get than with the 6.5X.
people will pretty much always feel like the higher gain on an amp is more dynamic, clearer, with better bass etc. that's because louder sounds better and very different indeed both in precision and in frequency response. but it's only a psycho acoustic trick and the signal is just louder. (or one resistor is badly soldered? you have the distortion in both ears?)
 
in effect it is always better to use the lower gain if it's loud enough. and that on any amp on the planet. for the O2 specifically, as the only weak spot for this amp is the channel imbalance at low volume, using the lower gain let you turn the knob out of that imbalance zone.
 
  My audio is set at 15/100 on the PC, 3 o'clock on the volume dial at 1x gain with the Q701's paired with ODAC/O2.
Once in a while I double the windows volume to 30/100 if I want to have "fun"

as advised, you should get that windozz volume up.
15% on my computer(win 7) is about -25DB(at least that's what windows is saying ^_^).
so with that setting, you're crushing(understand removing) the quietest 25DB of your track.
objectively, if your media player is set to output music in 24bit as advised by nwavguy, it doesn't matter and you can keep doing it as there will be no audible difference. 24bit being effectively maybe 20bit or 19bit so 20*6=120DB or 114DB. worst case scenario 114-25=89DB of music, that is good enough for audible perfection.
but then don't mess too much with replaygain as it's gonna remove even more DB. same for some EQ, so you can see how it is best to have some room. lower your O2 level a little and rise your windows level, the only limit to that should be widowzzz and your media player at 100%, or audible channel imbalance from the O2.
I spend most of my time at 9 o'clock and then fine tune with foobar.
 
Aug 19, 2014 at 5:51 PM Post #3,370 of 5,671
 
  Yup I see that as well at 100% but only with really hot songs.  If you use replay gain in JRiver this can be pretty well eliminated even with the player at 100%.
   
  Alex

 


I wonder if that's the case in foobar as well.


 so yes it happens but I can't remember if I was using wasapi when that clipping occured on some very very rare songs. I have something like -3DB on the preamp stuff of foobar just for that and since then I never experienced it again.
 
Aug 19, 2014 at 6:39 PM Post #3,371 of 5,671
 so yes it happens but I can't remember if I was using wasapi when that clipping occured on some very very rare songs. I have something like -3DB on the preamp stuff of foobar just for that and since then I never experienced it again.

 


Can you link to the article nwavguy wrote about windows volume and setting the output to 24bit? This dB talk is really confusing to me, but my understanding from this thread (http://bit.ly/1tffqzj) was that replaygain didn't audibly degrade sound quality, likewise with foobar's volume control.

EDIT: Btw is anyone else unable to get the BBCode editor to appear? This Rich Text editor is killing me and my posts.
 
Aug 19, 2014 at 7:50 PM Post #3,372 of 5,671
I am going to be testing my setup with the O2 amp at the lower gain setting. I am not sure if it will be lough enough since my foobar2000 settings lower the loudness by -13 Db due to crackling sounds (that is properly from the CD source themselves due to inadequate mastering).
 
But I do not want to blow out my headphone drivers so I have to test it out on different volume settings.
 
Aug 19, 2014 at 8:31 PM Post #3,373 of 5,671
 
 
 so yes it happens but I can't remember if I was using wasapi when that clipping occured on some very very rare songs. I have something like -3DB on the preamp stuff of foobar just for that and since then I never experienced it again.

 


Can you link to the article nwavguy wrote about windows volume and setting the output to 24bit? This dB talk is really confusing to me, but my understanding from this thread (http://bit.ly/1tffqzj) was that replaygain didn't audibly degrade sound quality, likewise with foobar's volume control.

EDIT: Btw is anyone else unable to get the BBCode editor to appear? This Rich Text editor is killing me and my posts.


like I said, individually nothing really matters. all we have to keep in mind is that we have a fixed available dynamic range and a track that need to fit in it. there are a few things that might lower that DR. so if we abuse them all, we can end up with very little left on the digital side. as long as you control what you're doing, there is no reason to be afraid of using anything, be it EQ, replaygain or foobar/windows volume control.
imagine someone using a powerhouse amp with high gain and sensitive headphones. let's imagine the amp has huge channel imbalance so he can't really use the amp knob. he will remove say 50DB with foobar to get the right sound level for normal listening. if the tracks already have replaygain you can pretty much remove 10db for some files. if he stays on 16bit that's 96DB available, meaning that he will at best hear 96-50-10=36DB of the loudest part of each tracks. not cool if you like classical or jazz.
 
about using 24bit for the odac, I don't remember where I saw that, but it's a real nobrainer. there are no negative side effect to outputting 16/44 tracks as 24/44(and let the DAC oversample when it needs to). it's just adding some zeros at the end of each sample, the original sample is kept untouched. so then if you digitally lower your volume too much for some reasons, then instead of starting to destroy the quietest sounds of the track, foobar will use those bits to write the music and you'll lose no data from the track(up to a point).
so if it's needed you have it, if it's not needed it's just a bunch of zeros that the DAC will note use. I would advise everybody to use asio or wasapi and select 24bit as output format. that way you can fine tune your system's loudness on the computer without having to think about it too much.
I wouldn't pretend anything about sample rate, but bit depth is actually very straightforward.
 
Aug 19, 2014 at 8:46 PM Post #3,374 of 5,671
like I said, individually nothing really matters. all we have to keep in mind is that we have a fixed available dynamic range and a track that need to fit in it. there are a few things that might lower that DR. so if we abuse them all, we can end up with very little left on the digital side. as long as you control what you're doing, there is no reason to be afraid of using anything, be it EQ, replaygain or foobar/windows volume control.
imagine someone using a powerhouse amp with high gain and sensitive headphones. let's imagine the amp has huge channel imbalance so he can't really use the amp knob. he will remove say 50DB with foobar to get the right sound level for normal listening. if the tracks already have replaygain you can pretty much remove 10db for some files. if he stays on 16bit that's 96DB available, meaning that he will at best hear 96-50-10=36DB of the loudest part of each tracks. not cool if you like classical or jazz.

about using 24bit for the odac, I don't remember where I saw that, but it's a real nobrainer. there are no negative side effect to outputting 16/44 tracks as 24/44(and let the DAC oversample when it needs to). it's just adding some zeros at the end of each sample, the original sample is kept untouched. so then if you digitally lower your volume too much for some reasons, then instead of starting to destroy the quietest sounds of the track, foobar will use those bits to write the music and you'll lose no data from the track(up to a point).
so if it's needed you have it, if it's not needed it's just a bunch of zeros that the DAC will note use.


Ah, I see. Makes sense.

I would advise everybody to use asio or wasapi and select 24bit as output format. that way you can fine tune your system's loudness on the computer without having to think about it too much.


I thought ASIO and WASAPI bypassed the system mixer, so it wouldn't matter what output format/loudness you chose?
 
Aug 20, 2014 at 5:46 AM Post #3,375 of 5,671
 
I would advise everybody to use asio or wasapi and select 24bit as output format. that way you can fine tune your system's loudness on the computer without having to think about it too much.


I thought ASIO and WASAPI bypassed the system mixer, so it wouldn't matter what output format/loudness you chose?

I use wasapi(only because It shows each outputs in foobar separatly so I can use shortcut keys to switch between them. no audio quality reason) and it let you pick the output bitdepth so it's your choice and that choice. on asio I seem to remember that it will select the highest possible bitdepth for your DAC/soundcard (showing that at least asio and me are on the same page for audio output bitdepth ^_^).
asio and wasapi are here to bypass windows mixer and whatever thx surround stuff that may be working in the background of your soundcard, you're perfectly right about that. but your audio player can still do whatever it likes. volume, EQ, replaygain are all working within the audio player effectively ruining the bit perfect concept.
 

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