[Zac.R]

To the OP: everything becomes 'endlessly complex' if you try to analyse everything at once.

Read ROBSCIX's posts - the answer to your question is very straight forward. Set all volume levels on your PC to 100%. Use your external amp for your volume contol. Least noise, greatest signal. Done.

 

[JonathanHF]

One night when I listened to songs through my headphone with external soundcard I found that the sound was rather muddy and veiled.
Later I found that the volume on my laptop was set to 80 instead of 100.
After pulling it to 100 the sound opened up and everything became more transparent.
To me it is obvious what the volume control can do to the sound.

 

[anwaypasible]

have any of you taken into consideration the 'coloring' effect that the amp has at different volume/energy output levels??!

i understand what was said in this post and i am not saying what was said is impossible.

however, i would like to steer the conversation to the other side of the story to further complete this scenario.

do i know exactly how well the windows volume sliders work? no
do i know for certain that the soundcard provided software has a better volume control? no
(which is reason ONE to put all software volume knobs at 100%)

now listen,
i have moved into an apartment and i've calibrated my speakers numerous times for this room (mostly because i started with one set of shelf speakers then moved to my 12 inch woofer towers, then swapped woofers, then pulled out the midranges and set them on top of the box in free-air with the other hole covered up/sealed)
so if you count - there are 4 times that i have calibrated this room.
but again yesterday i opened up my windows and pulled up the blinds for spring weather and had to re-calibrate again.

but specifically, if you were paying attention to what i said above, i calibrated the original 12inch woofers TWICE.
once with my amp/receiver at 56% volume - and then again with the receiver volume at 70%

the reason i re-calibrated with the receiver at 70% volume is because my 12's didnt have the power going to them to get down into the 20hz area.
after turning the receiver up, the EQ sliders are different since there are different needs based on the power going to the speakers.
the sound is much more pleasant at low volume now.. i am able to hear well down to 30hz and below.

that is key number one that needs to be said.
and thus with that said, there are other important issues that need to be addressed.
resistors and capacitors have different characteristics when different voltages are applied to them.
(i'm sure with time and the education of mankind, the differences will be much less if existing at all.. with a premium price of course)

so by adjusting the analog volume on the amplifier, you are really taking a chance at 'coloring' your precious flat-response curve.

it is much easier to digitally address decibel levels of sound than compared to an analog method.
remember folks, there were skeptics and engineers who would tease and promote the fact that old EQ's would 'color' the sound while boosting or decreasing SPL because of the components involved.

it is true that some 'bits' of music data may be thrown out when digitally turning the volume down.
very soft sounds would be lost at first chance because they are already hardly audible.

and then you should take a second look and realize that 'digitally' the sound wave is only stretched or shrinked and sent through an amplifiers components to make the speaker cones move more or less (which is the sole reason for sound pressure levels)
it is possible to send all data to the speaker cone regardless of how audible the sound is.. but the speaker may have a harder time producing anything the human ear can pick up on at lower cone travel. (and some speakers will have a harder time producing any sound when the cone travel is much.. a speaker quality rather than sound data problem)

i'm also quite sure that there are algorithms that exist to keep all subtle sounds at a perceptual level so they can be heard rather than lost during transmission between the time the speaker cone plays it and reaching the eardrum.
(of course this is one reason why headphones are so spectacular for use)

anyways.. you will know what i am talking about when you have a speaker or headphone that produces no bass at low volume and must be turned up for any depth into the lower frequency range.
then there are other speakers that can still reproduce bass notes regardless of power applied to the speaker motor.
very soft and subtle bass that comes through as a whisper (including vocals and tremor) or very loud notes that can be heard by someone sitting on the couch with you.

i leave my analog volumes alone and use software volume controls.
although i am not certain that creative hasnt tied all windows mixer volume controls to their own volume design incorporated in the software/drivers.
maybe i am turning the analog stages of the soundcard down without knowing it.. but my amp/receiver is always at 70% volume.
it would be better to keep my analog components on the soundcard at a constant level just like keeping the receiver at constant level so there are no coloration problems from a varying voltage being applied to components on the circuit board.
the dynamics would also be improved because the extra power is there when needed rather than sucking for more power from the voltage regulator (which is designed to say 'no' in the first place)

i will say that i'm sure there are audio players out there that have very poor volume controls that really change the sound when sliding the control across the scale of useability.
but i trust the big-name software and my ears, which tell me that the decibel levels are always there but the only thing missing is SPL.

there is a huge difference between hearing something audible with little/no air between the speaker and eardrum compared to hearing and FEELING the air around you sway and phase to a rhythm as intended (usually be an entire wall of speakers properly tuned to the walls of the room)

indeed there are audio 'effects' that try to simulate the sound changes when the air in the room sways from one side to the other, and i'm certain that with practice these false 're-creations' will be audibly heard as truth/fact.
the only difference is the wind that sweeps your hair from side to side (which is really more than mind-boggling)

i had my speakers roughly 3ft away from each ear and then put one speaker in each corner of the room.
after putting the speakers in each corner of the room and calibrating my timing alignments in a THX console, i have now been able to feel changes in the sound pressure levels put there by the original producers.
and let me tell you, there are quite a few songs that have hidden SPL rhythms that go completely unaware to headphone users.

especially since, when done properly, there is nothing audible in difference.. but merely a change in wind/air and the phase/direction that air moves.

think of it like this.. sure you can blow your cheeks out with air (both at the same time/pressure)
but you can also blow up one cheek and swish the air from side to side or any combination inbetween.

while it may be possible for headphones with large xmax to provide your ears with the same 'air-stunts'
the difference is when you feel those 'air-stunts' on your entire body (or in your entire room as well).

with the economy in mind.. i really dont think there are many headphones out there with that can deliver the xmax on the cone while keeping up with all the crisp details in the audio.
because once you get the crisp details in the audio taken care of.. the next step is to add xmax AND KEEP THOSE DETAILS which is rather hard to do (and comes with a premium price tag)

ya know.. you can throw out 40hz from some tower speakers, but if you throw out the same 40hz out of phase so it isnt heard / and then take the vacuumed space that you are in (the air inside the room) and create a 'tide' like an ocean or lake.. that 'tide' will be considered an optimisation because instead of a giant room full of sound (lots of echos) the sound now is perceived as if there were zero echos thanks to the crest or ripple of the wave hitting your ears and nothing else (i forgot what the tip of the wave is called).

audio has come a long long long way.
back when castles were the town and speakers played only voltages applied from cranking generators that were made to play with the coil inside the magnet at a CONSTANT rate rather than a very mild back and forth.

funny yes, but the transducer did come before the electric generator.
somebody probably was going through life and felt as though their 'battery-pack' was attached with a very long cord.
so long in-fact that the cord had to be wound up in a spool to be organized.
and with all the kinetic energy between humans, it only seemed right to use magnets in conjunction with the concoction.

anyways..
just remember that a 5hz sinewave is exactly that.. a wave that moves the cone.
the higher the wave the more the speakers move.. and stretching or shrinking the wave digitally is very very easy.
the scenario only gets more complex from here if you allow it to be.

 

[tuoppi]

Simply put your computer doesn't send the volume information separately. For example when the music is 16-bit, only the 16-bits are sent out. At 100% volume the music fills the 16-bits like it normally does, there is no change. When you lower the volume the music is sent out with 16-bits, but it does not use all 16-bits (or rather all the combinations of bits available), it uses less and the music had to have been changed.

We could simplify this to make it easier to understand. Let's assume that your music can have 8 different levels. Now at 100% volume the music can use any numbers between 1 and 8. But when you lower the volume your computer tells the music not to use 7 or 8, and the music is changed so that it fits between 1 and 6. In the latter case 8 levels are still sent out, but 7 and 8 are not used.

 

[anwaypasible]

Quote:

Originally Posted by tuoppi /img/forum/go_quote.gif Simply put your computer doesn't send the volume information separately. For example when the music is 16-bit, only the 16-bits are sent out. At 100% volume the music fills the 16-bits like it normally does, there is no change. When you lower the volume the music is sent out with 16-bits, but it does not use all 16-bits (or rather all the combinations of bits available), it uses less and the music had to have been changed. We could simplify this to make it easier to understand. Let's assume that your music can have 8 different levels. Now at 100% volume the music can use any numbers between 1 and 8. But when you lower the volume your computer tells the music not to use 7 or 8, and the music is changed so that it fits between 1 and 6. In the latter case 8 levels are still sent out, but 7 and 8 are not used.

it takes a lot of signal to completely fill the 16bit architecture.. and once the 16bits is full, the resulting sound would be stupid and extremely fatiguing.
(picture four songs playing at once through the same speakers)
levels 9-20 would be 24bit (just to add to what was said in the quote above)

there are four different ways to go to start.
1. the highest allowed decibel
2. the lowest allowed decibel (-140dB)
3. the lowest frequency allowed
4. the highest frequency allowed

the above four help dictate what bit-depth is used.
now the xfi 24bit crystalizer does an amazing thing when used properly.
say for-instance you have a 16bit song that has some peaks that fall out of the 16bit range (would ruin the entire track with processing to fix)

nowadays producers can take those peaks and pull them down until they reach a decibel level that falls within the 16bit depth.
then the 24bit crystalizer takes those 'squished' peaks and puts them right back where they were before audible output.

it is really a compression and de-compression method for 24bit sound.
but thinking logically - 24bit crystalizer is only for 16bit audio that has some peaks into the next bit depth.
-- eventually the 16bit track will fill up and there will be no room for any data, and this is with all sounds that fell into the next bit-depth category ( + the other data that resides in the 16bit depth) --

that is why you cant take a 24bit track and compress it into 16bit and then unpack it to 24bit again.
there isnt enough room for the full 24bit track.

think of it like 16bit music 'over-clocked'

decibel peaks in both directions are easy.. but adding frequency response is much much more complex (approaching impossible)

very pathetic 24bit audio is possible with a 16bit depth transfer.
(like only 20% of what 24bit audio is capable of)

but once music starts going to 24bit as normal.. there will be peaks into the next bit depth and the whole process will be reproduced again.

 

[tuoppi]

Quote:

Originally Posted by anwaypasible /img/forum/go_quote.gif it takes a lot of signal to completely fill the 16bit architecture.. and once the 16bits is full, the resulting sound would be stupid and extremely fatiguing. (picture four songs playing at once through the same speakers) levels 9-20 would be 24bit (just to add to what was said in the quote above)

I didn't mean filled like that. I meant that the music can use any bit combinations possible with 16 bits. With less than 100% volume it can't use all the possible combinations.

PS. I don't know what you are on about, I am just trying to explain digital volume.

 

[anwaypasible]

and that is how VST plugins work when manipulating audio within whatever bit-depth the plugin is designed for (or aimed at).

we are on about audio mastering within a certain bit-depth with this post.

for example.. dragging harmonics from the next bit-depth and making them fit in the lower bit-depth (and then arranging those harmonics in conjunction with the rest of the audio in the bit-depth that was already there).

lots of lazyness makes the audio sound fatiguing because so much of the track is a the same decibel level and the data hasnt been flourishly arranged within the bit-depth.

(and other times there simply isnt enough room because proper trimming hasnt taken place with the use of certain - less dynamic microphones)

 

[joashtzh]

this is my computer audio set-up:
 
foobar w/ASIO4all (no dsps) > nuforce icon HD dac/amp > UM3x
 
As you all know UM3x is a really sensitive IEM, so usually i have to turn the amp volume pot to just above 9 o'clock, which is just barely above the channel imbalance range.
 
However, this is still way too loud so i either have to reduce my system volume or reduce volume on foobar software.
 
My question is, in theory, which method of volume reduction is less degrading to the digital usb out signal? or are there other better ways that allow me to listen to my UM3x w/o harming my eardrums?

 

[Battou62]

This is something I have been trying to figure out with my X-Fi HD USB. The volume knob on the unit directly moves the volume slider in windows. I am driving a pair of JVC HARX900's and the comfortable listening range is somehwere between 20-25% of max volume. I have been sending Creative tech support emails to try to sort out if this is casuing any audio degradation but have not been able to get a solid answer.
 

 

[Brent Hutto]

Look up "gain staging" in Wikipedia or somewhere...

If you take the necessary voltage to play music on your IEM/headphone at proper volume, divide that by the voltage emanating from your computer (with the "digital volumes" set to full-on) and express the result in decibels, that is the correct gain for your headphone amp. Too much is bad, too little is bad, there is a fairly narrow range for correct gain.

If the correct headphone amp gain can not be achieved without channel imbalance or other side effects, your setup is broken. You need a different amp, different headphone, different source. Working around it by lowering the "digital volume" settings on the computer is cutting off your nose to spite your face.

You're buying the best equipment you can afford and using a headphone amp to improve the sound quality. Screwing up the sound coming out the computer with the "digital volume" controls will degrade the sound quality substantially. You gotta ask yourself why you're messing with the amp in the first place if it requires to you throw away a major part of the information making up the musical signal in the first place. It's like paying your boss $10,000 to give you a $5,000 pay raise.

 

[JRG1990]

Quote:

this is my computer audio set-up:
 
foobar w/ASIO4all (no dsps) > nuforce icon HD dac/amp > UM3x
 
As you all know UM3x is a really sensitive IEM, so usually i have to turn the amp volume pot to just above 9 o'clock, which is just barely above the channel imbalance range.
 
However, this is still way too loud so i either have to reduce my system volume or reduce volume on foobar software.
 
My question is, in theory, which method of volume reduction is less degrading to the digital usb out signal? or are there other better ways that allow me to listen to my UM3x w/o harming my eardrums?

 
Yea you have to choose between digital (lesser quality) or analog (channel imbalance) , i think nuforce uses so pretty poor quality pots the 1 on my fiio comes out of channel imbalance  just below 9 o'clock ,..
 

 

[FDS]

Quote:

As you all know UM3x is a really sensitive IEM, so usually i have to turn the amp volume pot to just above 9 o'clock, which is just barely above the channel imbalance range.  
However, this is still way too loud so i either have to reduce my system volume or reduce volume on foobar software.
 
My question is, in theory, which method of volume reduction is less degrading to the digital usb out signal? or are there other better ways that allow me to listen to my UM3x w/o harming my eardrums?
 

Quote:

Originally Posted by JRG1990 /img/forum/go_quote.gif
 
Yea you have to choose between digital (lesser quality) or analog (channel imbalance) , i think nuforce uses so pretty poor quality pots the 1 on my fiio comes out of channel imbalance  just below 9 o'clock ,..

I think the hdp was designed more for high impedance headphones. Though nuforce claims it's good for iems too, I would imagine it's very difficult if not impossible to make a headphone amp that's compatible with everything from HD800s to extremely sensitive iems. As Brent mentioned, it's important to match components in your setup and think about the system as a whole.
 
However, it seems that sometimes this problem arises even when the headphone amp and headphones go well together. For example, I had the same problem with my zero dac + K701. Obviously the K701 take quite a lot of juice to drive. The last thing I suspected was the zero's headphone amp to be too powerful for them. Using WASAPI, and having all software volume controls maxed, the output would be much too loud for comfortable listening. I would have to turn the volume knob on the zero way down to almost off. The zero, like most other analog volume pots, had very bad channel imbalance below about 12 o'clock. This made it impossible to listen to with software volume controls maxed. I really had no other option than to reduce windows control by 50% or use the -dB in foobar until I got to the point where I could move the volume knob past the channel imbalance.
 
This is the problem I think the OP and several other posters in this thread, including me, are trying to address. A lot of the previous posters have neglected to answer "what should one do when the output is just too loud with software volume controls maxed?"
 
Leeperry mentioned possibly using a higher quality volume pot, but obviously this isn't practical for most of us. The other option is just using a different headphone / amp combination for a better pairing. Is there anything we can do that doesn't involve an equipment upgrade/change?
Quote:

it's kind of a mixed bad sometimes though, because high efficiency headphones usually don't allow you to go over 9 O'clock on most amps...and only the most expensive pots don't have major stereo imbalance issues <9 O'Clock.

 

 

[leeperry]

actually you really don't want to output 100% volume from the computer for a very good reason: http://www.hometracked.com/2007/11/08/prevent-intersample-peaks/

 

[FDS]

Quote:

actually you really don't want to output 100% volume from the computer for a very good reason: http://www.hometracked.com/2007/11/08/prevent-intersample-peaks/

Um, ok, so now you're saying that the recommendation to put all software volume controls at 100% that we have seen posted over and over again on this forum is wrong? I thought lowering software volume controls meant throwing out bits? That's what previous posts in this thread have been pointing out.
Quote:

The reason is that using software to reduce the volume accomplishes this goal by throwing away bits. So you actually get lower resolution than you would if you were to have the software at 100%, and use the amps analog volume control to adjust the volume.

Quote:

To maintain the audio data of the played back audio file. Simple as that.
Since digital volume controls are lossy.

 

 

[leeperry]

 

Um, ok, so now you're saying that the recommendation to put all software volume controls at 100% that we have seen posted over and over again on this forum is wrong? I thought lowering software volume controls meant throwing out bits?
 

It's already been discussed(and proven) here: http://www.head-fi.org/forum/thread/542358/im-new#post_7315509
 
A worst case scenario will require -6dB attenuation, and 6dB equal to 1bit. I'm sure you can live w/ 15bit...15*6=90dB. I don't think many ppl would be crazy enough to listen to headphones at >90dB volume
 
The added value is that the DAC stage will NEVER clip, even for hardcore loudness war material.