Confused w/ my setup: Compass, AH-D7000, and iTunes
Jul 7, 2009 at 3:27 AM Thread Starter Post #1 of 11

estwd

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Hi all,

Here is my setup:

Apple Lossless via itunes ->usb out -> Audio-gd Compass usb in to headphone out ->Denon AH-D7000.

Here is my confusion. Why am I able to change the volume in iTunes and here it via the headphones? Isn't my connection digital from my computer to my dac/amp? Shouldn't the amp in the Compass be the only way to affect volume? But sure enough, if you slide the volume slider in iTunes the volume goes up and down. Yet if I change the volume of my sound card nothing happens (as should be expected because the audio is set to output via usb digitally).

I only noticed this when I set my iphone up as an iTunes remote and was able to change volume even though I was listening via the Compass which only had a usb cable connection.

BTW, my Compass now all of the sudden sounds flat after I set the iphone up as a remote to control my pc's iTunes.

I have 1000 hours on my Compass so it isn't a burn in issue,

Please enlighten me.

estwd
 
Jul 7, 2009 at 4:06 AM Post #2 of 11
If it is not USB out via ASIO, I beleive that the volume control is not bypassed. It is not a DAC thing, it is that the data stream going to the DAC says "scale me down by X amount" <- simplified terms, obviously
smily_headphones1.gif
.

-Nkk
 
Jul 7, 2009 at 9:52 AM Post #3 of 11
Hmm, well, digital volume control?
wink.gif
You want to keep iTunes' slider all the way up.

What kind of "flat" are we talking about? Take a look to see whether iTunes' Sound Check is engaged. Unless you have replaced it with an MP3gain implementation, that will likely not sound the best.
 
Jul 7, 2009 at 11:28 AM Post #4 of 11
How do I get iTunes to stop scaling back the digital signal? Ultimately I am going to move the Compass downstairs and connect it to iTunes via an optical cable connected to an Apple Airport Express. Will this fix the volume issue and dumbing down of the digital signal?

Quote:

Originally Posted by nkk /img/forum/go_quote.gif
If it is not USB out via ASIO, I beleive that the volume control is not bypassed. It is not a DAC thing, it is that the data stream going to the DAC says "scale me down by X amount" <- simplified terms, obviously
smily_headphones1.gif
.

-Nkk



 
Jul 7, 2009 at 11:42 AM Post #5 of 11
iTunes have a built in digital volume control, which do not disable when using USB out.
Just leave the slider at max (100%) and it should be out-of-the-way.

Quote:

Originally Posted by estwd /img/forum/go_quote.gif
How do I get iTunes to stop scaling back the digital signal? Ultimately I am going to move the Compass downstairs and connect it to iTunes via an optical cable connected to an Apple Airport Express. Will this fix the volume issue and dumbing down of the digital signal?


Yes, when using AirTunes you have the option in iTunes to disable its built in volume control.
 
Jul 7, 2009 at 12:44 PM Post #6 of 11
I can understand what digital audio control is - in theory, but can't wrap my brain around how it works technically. How do you digitally increase the volume of a bunch of zeros and ones? Wouldn't the signal have to be analog to hear any change in volume?


Quote:

Originally Posted by krmathis /img/forum/go_quote.gif
iTunes have a built in digital volume control, which do not disable when using USB out.
Just leave the slider at max (100%) and it should be out-of-the-way.


Yes, when using AirTunes you have the option in iTunes to disable its built in volume control.



 
Jul 7, 2009 at 7:40 PM Post #7 of 11
Also, how do I know if/prevent iTunes from dumbing down my digital signal when outputting via USB or toslink to the Compass. When using toslink I'll be using the toslink mini connector on the Apple Airport Express.

Do others think I should just keep the digital volume control maxed out on iTunes when sending the digital signal to the Compass? Will this distort the sound from the Compass?

Please advise

estwd
 
Jul 7, 2009 at 7:49 PM Post #8 of 11
Quote:

Originally Posted by estwd /img/forum/go_quote.gif
I can understand what digital audio control is - in theory, but can't wrap my brain around how it works technically. How do you digitally increase the volume of a bunch of zeros and ones? Wouldn't the signal have to be analog to hear any change in volume?


interesting question, i am also don't understand how it works exactly, but there is such a thing as "digital volume control" and i guess it means that one's and zero's can be turned into volume also in the same way that they can be turned into tones and notes...you know, more bass treble.
but this is rather an interesting subject.

just put it on 100% like others said..all the times when i asked the same question i got the same answer, so i guess it is true to put it on full volume. it works like an attenuator i guess...only turning down the volume and not raising it up,so to put it on 100% is the logical thing.
by the way,if it is a digital volume control i don't think that it can downgrade your sound in any way (unlike an analog volume that can degrade the sound if it is not on "line-level" output) but don't count me for that, let's see what others have to say about this.
 
Jul 7, 2009 at 9:56 PM Post #9 of 11
Quote:

Originally Posted by estwd /img/forum/go_quote.gif
I can understand what digital audio control is - in theory, but can't wrap my brain around how it works technically. How do you digitally increase the volume of a bunch of zeros and ones? Wouldn't the signal have to be analog to hear any change in volume?


The 1s and 0s are volume, for all practical purposes, and you can only lower them without risk of clipping [size=xx-small](analog volume controls also tend to only lower volume)[/size]. A useful audio signal is a bunch of changes of amplitude at certain rates, whether digital or analog. PCM audio at 44.1kHz has the amplitude at 44,100 different points during each second. In normal (linear) PCM, they are values of the amplitude from full scale at the current time, like 6345/65535, or 81/65535. The DAC stitches them together and smooths it out to an analog waveform.

http://en.wikipedia.org/wiki/File:pcm.svg
The value is the left corner of each step in the shaded area of the graph. If you halve the value [size=x-small]([size=xx-small]y'=int((y-7)/2)+7[/size]?)[/size], you get a wave of the same frequency, but half the amplitude: digital volume control. Thing is, you also lose one bit of resolution for each half, and some volume controls can cause audible artifacts [size=xx-small](can, not will)[/size].

16-bit audio has a minimum amplitude that is above what many modern circuits can manage for SNR, and only just at the limits of human hearing [size=xx-small](so if the digital volume is set very low, but you listen at normal volumes, you can lose plainly audible detail)[/size]. So, you leave that at maximum, and attenuate out in the analog realm, for minimum potential loss.
 
Jul 8, 2009 at 12:51 PM Post #11 of 11
Amazing clarification.

Thank you 44,100 times.

Will my Apple Airport Express with mini toslink to my Compass cause any bottleneck in the process?

estwd

Quote:

Originally Posted by cerbie /img/forum/go_quote.gif
The 1s and 0s are volume, for all practical purposes, and you can only lower them without risk of clipping [size=xx-small](analog volume controls also tend to only lower volume)[/size]. A useful audio signal is a bunch of changes of amplitude at certain rates, whether digital or analog. PCM audio at 44.1kHz has the amplitude at 44,100 different points during each second. In normal (linear) PCM, they are values of the amplitude from full scale at the current time, like 6345/65535, or 81/65535. The DAC stitches them together and smooths it out to an analog waveform.

http://en.wikipedia.org/wiki/File:pcm.svg
The value is the left corner of each step in the shaded area of the graph. If you halve the value [size=x-small]([size=xx-small]y'=int((y-7)/2)+7[/size]?)[/size], you get a wave of the same frequency, but half the amplitude: digital volume control. Thing is, you also lose one bit of resolution for each half, and some volume controls can cause audible artifacts [size=xx-small](can, not will)[/size].

16-bit audio has a minimum amplitude that is above what many modern circuits can manage for SNR, and only just at the limits of human hearing [size=xx-small](so if the digital volume is set very low, but you listen at normal volumes, you can lose plainly audible detail)[/size]. So, you leave that at maximum, and attenuate out in the analog realm, for minimum potential loss.



 

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