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[PrinterPrinter]

Hello,
I love MC and it generally works fine.

I have a unique problem in my set up and wanted your thoughts.

My dedicated HiFi set up is built around a fan-less computer running MC (Streaming music via Wifi from my main computer also running (MC) where all the music files are stored. I don't have any corrections running other than volume leveling.

It's then connected via Toslink to a chord Hugo DAC that goes into a Classe Cap2100 amp in power-amp mode. It all sounds great but my problem is that I usually listen at reasonable volumes (what most of you would probably call low) and the Classe has a very strong gain in power-amp mode, so basically I'm running the Hugo at it's lowest volume putting out around 0.1v into the Classe via RCAs.

I'm using the JRMCs internal volume control - so I can change it from my iPhone via JRemote.

I've been trying to increase the Hugo's output volume without blowing my speakers (and neighbours) away and came up with an idea of running JMRCs internal volume very low (around 35 out of 100) - at which case I can increase the Hugo's output to around 1v and the music sounds fuller I think - while keeping the overall volume levels sane.

I have tried using the amp in integrated mode or adding an offset on that line but the sounds is noticeably constrained and less open.
My questions are as follows:
1. Is running the volume on MC at 35 limiting or hurting SQ in any way? I thought because it's all digital it wouldn't be the case.

2. Limiting the volume - because now potentially someone can turn it up to 80 and blow the house by mistake I limited it to 50, though I wonder if I can somehow retain the flexebility of having the volume go up to 100 in the app but in reality not going over 50 - some sort of an offset?
3. Lastly, in the future I would like to use the sophisticated room corrections functions of MC - would using a low volume somehow limit its bandwidth?

Thanks!

[mwillems]

I've been trying to increase the Hugo's output volume without blowing my speakers (and neighbours) away and came up with an idea of running JMRCs internal volume very low (around 35 out of 100) - at which case I can increase the Hugo's output to around 1v and the music sounds fuller I think - while keeping the overall volume levels sane.

I have tried using the amp in integrated mode or adding an offset on that line but the sounds is noticeably constrained and less open.
My questions are as follows:
1. Is running the volume on MC at 35 limiting or hurting SQ in any way? I thought because it's all digital it wouldn't be the case.

Significant digital attenuation (theoretically) reduces the dynamic range of the output, but I don't think it's really relevant to normal listening.  A full scale 24-bit output has about 144 dB of dynamic range (plus a bit more for dither).  By using MC at 35% (-32.5dB) you're reducing the effective dynamic range of a 24-bit output to 111.5 dB (plus a bit for dither).

111.5 dB is still quite a lot, however.  16-bit audio (which is what's on regular CDs) only has 96dB of dynamic range.  Because MC uses the highest bitdepth available on your device by default, if your device supports 24-bit output, you can reproduce regular CD audio without even a theoretical loss of dynamic range. What's more human hearing (in non-laboratory settings) only has a dynamic range of about 100dB anyway.

If your device only supports 16-bit outputs, you might be actually reducing the dynamic range to the point that it could be audible: 96dB - 32.5 is 63.5 dB, and it's possible that you might be able to hear things that much quieter than the rest of the music.  Possible, but not likely because most people have a hard time hearing sounds that are more than 40dB quieter than other sounds at the same time.  But it's at least plausible that a person could hear sounds 60dB down.  But I wouldn't worry about it very much.  We're talking about very, very quiet sounds.

My normal listening volume in MC is 20% (for similar reasons to yours), and I have no concerns that I'm missing anything (although I have a 24-bit device and mostly listen to CD audio).

2. Limiting the volume - because now potentially someone can turn it up to 80 and blow the house by mistake I limited it to 50, though I wonder if I can somehow retain the flexebility of having the volume go up to 100 in the app but in reality not going over 50 - some sort of an offset?

You can reduce the volume of all channels in parametric equalizer.  That would have the effect you want.

3. Lastly, in the future I would like to use the sophisticated room corrections functions of MC - would using a low volume somehow limit its bandwidth?

Just the opposite actually.  MC's DSP path is 64 bit so it can easily handle very quiet sounds, and having lower internal volume actually provides additional headroom for DSP.  Since most digital music is mastered so that it peaks at 0dBFS, adding any boost using DSP to a full-scale signal will cause clipping at some point.  By reducing internal volume in MC, you give MC's DSP the headroom to boost certain freqeuncies if necessary without driving everything into clipping.  In your case, quite a bit of headroom!

[PrinterPrinter]

Thanks Mate that's very helpful and very interesting.
My DAC works at 32Bit (and that's what the audio path shows in MC).

One last question about limiting the volume,
I tried using the parametric EQ with 'Adjust the volume' by a -24Db - it does that but now the audio path icon isn't blue - is teh SQ compromised somehow? Is there a better way of doing this?

Lastly, I noticed there's an option for 'internal volume reference level' - it's currently set to 100 - what does it control and what happens if I reduce it by half?

Many thanks!

[~OHM~]

Just the opposite actually.  MC's DSP path is 64 bit so it can easily handle very quiet sounds, and having lower internal volume actually provides additional headroom for DSP.  Since most digital music is mastered so that it peaks at 0dBFS, adding any boost using DSP to a full-scale signal will cause clipping at some point.  By reducing internal volume in MC, you give MC's DSP the headroom to boost certain freqeuncies if necessary without driving everything into clipping.  In your case, quite a bit of headroom!

I always wondered about this...thanks for clarifying

[Matt]

Lastly, I noticed there's an option for 'internal volume reference level' - it's currently set to 100 - what does it control and what happens if I reduce it by half?

The reference level sets the volume point for using the Loudness feature.

A little more here:
http://wiki.jriver.com/index.php/Volume

[~OHM~]

Lastly, I noticed there's an option for 'internal volume reference level' - it's currently set to 100 - what does it control and what happens if I reduce it by half?

Many thanks!

That is what you are looking for...it controls the slider volume control on MC....setting it at your percentage will infact not allow anyone to up the volume by accident....also just below that one is the start up volume you could set it to match the first....

[mwillems]

it does that but now the audio path icon isn't blue - is teh SQ compromised somehow? Is there a better way of doing this?

The SQ is only "compromised" in the way I described above; you have 24dB less dynamic range than you otherwise would have, but for the reasons I mentioned, it's not likely to have a practical effect with a 24+ bit DAC (especially if you're listening to 16-bit music).

[Vincent Kars]

I wonder if this is about dynamic range.
Both digital and analog volume control affect the dynamic range.
In case of digital volume control, each reduction of 6 dB cuts of a bit.
If the volume reduction becomes substantial, you are losing resolution.

Anyway, it is something you can easily try.
Compare the digital volume control set to 100 with analog to low and visa versa.

The Classe has an option “input offset”
You might try setting it to its lowest value (reduced gain)

[mwillems]

Both digital and analog volume control affect the dynamic range.

How does an analog volume control affect dynamic range?  Digital affects dynamic range because it has a limited field of representation (bitdepth), so if you shift everything down, the quietest bits "fall off the bottom."  There isn't a similar limitation in analog.

Do you mean the (slight) addition to the noisefloor of the amp that the electronic components necessary for a volume control add?  I guess that could be conceptualized as dynamic range, but a well-designed volume control should contribute the same(or similar) Johnson noise regardless of its setting (so that ship has sailed once the device has a knob on it).

I ask this as someone who prefers to use digital volume controls in my own setup; I've just never heard an argument that analog volume controls affect dynamic range per se (except to the extent the physical control is a source of a noise).

[Vincent Kars]

Sorry, I mixed up loudness with dynamic range.

As far as I know, analog VC lowers the noise floor and digital doesn't
Anyway, what's you opinion about this article?
http://www.esstech.com/PDF/digital-vs-analog-volume-control.pdf

[mwillems]

I agree with most of their factual statements, and I (mostly) agree with their conclusions.  Their examples primarily use a 16-bit DAC, and as I noted above, with 16-bit DACs it's easy to run into (at least theoretical/measurable) issues.  As they note with a higher bitdepth (they use a 32-bit example), the problem is much less acute.

Their final conclusion is basically correct but forgets to add a few significant points for context:

Analog volume controls easily outperform digital, unless the
digital control has access to the data path of the DAC
(ie is internal to the DAC)
–
Exquisitely well designed analog volume controls can still beat
even the very best internal digital volume controls if they have a
lower noise floor than the DAC itself [emphasis added]

That statement is technically correct, but there are two caveats:

1) A well-designed analog volume control can always outperform a digital volume control, as they note "if they [the analog volume controls] have a lower noisefloor than the DAC itself."  That is correct, but it is non-trivial given that the noisefloor of the nicer modern DACs are essentially a few dB above the thermal noise of the resistors involved! I've encountered many analog volume controls over the years (even ones made by reputable  manufacturers), that were quite noisey.  Low-noise volume control design is not a dark art, but it's not guaranteed either.  So this point is technically correct in the abstract, but there's no guarantee that the volume knob on any particular piece of kit is going to be lower noise than a well-designed digital volume control (and it might well be quite a bit worse).

2) If you're reproducing 16-bit music on a 24-bit (or 32-bit) DAC, the "performance" of the volume control is entirely irrelevant unless it compromises the 96dB envelope of the original recording.  Put another way, even if you lose 40dB of dynamic range using the digital volume control on a 24-bit DAC, you still have 104dB of dynamic range, which is more than the source material.  Because there is no dynamic range to "lose," the performance of the volume control is a red herring.  With 24-bit music you may be discarding some very quiet bits, but whether bits that are effectively beyond the dynamic range of human hearing (outside of a lab) are important or not is something each of us will have to decide for ourselves.

For my part, 98% of my audio is redbook CD (i.e. 16-bit), and I'm not generally inclined to roll the dice on an analog control (my amps are block amps)

[Vincent Kars]

Thanks

[PrinterPrinter]

Thanks for the great advice.

A few last question:

1. I also use volume leveling which often runs at -14dB, together with the lower voume (to compensate for the amps gain - I'm looking at close to -45dB, taht woudl start to be noticable? Certaibly on 24Bit track or also on redbook too? If I turn volume leveling off than the music would just get louder and I'll need to lower the volume further... right?

2. The Chord Hugo claims to be a 32Bit Dac - but with 120Bit of dynamic range... http://chordelectronics.co.uk/products-info.asp?id=92 does this influence the above question about headroom? MC says it has a direct connection at 32Bit.


Thanks Guys!

[mwillems]

1. I also use volume leveling which often runs at -14dB, together with the lower voume (to compensate for the amps gain - I'm looking at close to -45dB, taht woudl start to be noticable? Certaibly on 24Bit track or also on redbook too? If I turn volume leveling off than the music would just get louder and I'll need to lower the volume further... right?

You have to add up the total attenuation, so in your case -45dB.  I think that level of attenuation is unlikely to be noticeable, but you would certainly be losing some dynamic range (necessarily) on 24 bit material.  For 16-bit, I think you'd still be fine (or at least, nearly fine; see below).

2. The Chord Hugo claims to be a 32Bit Dac - but with 120Bit of dynamic range... http://chordelectronics.co.uk/products-info.asp?id=92 does this influence the above question about headroom? MC says it has a direct connection at 32Bit.

That's something I glossed over above; a common issue is that, due to the noise generated by electronic components, no existing DACs actually can fully resolve 24bits of dynamic range (i.e. no DAC for sale has a dynamic range of 144dB or higher), so a 32-bit dac will not necessarily perform better than a 24 bit one (on that basis alone).  If your DAC has a 120dB dynamic range, then you have -24dB of attenuation before you theoretically compromise the 96dB envelope of 16 bit music.  But even if you do your full -45dB of attenuation, such that the effective dynamic range of your DAC becomes 75dB, any information that you're losing is 75dB quieter than the loudest part of the music, probably more than 65dB quieter than the average volume of the music (depending on the music, of course).  

But, the important thing to keep in mind is that noisefloor of an average home (when it's quiet) is between 30dB and 50dB.  Labs spend many thousands of dollars to construct "quiet rooms" that have noisefloors in the 20dB range, or anechoic chambers which might have even lower noisefloors.  So in a home setting, even with soundproofing, it's unlikely that the noisefloor is going to get much below 30dB.

So unless you routinely listen to music at 100+dB (or have an anechoic chamber in your house), the rate limiting factor is not your DAC, but the "dynamic range" of your room!  If you listen to music at 85dB (which is actually pretty loud in a home setting), the dynamic range of your room is probably about 45dB (maybe 55dB if you turn your ventilation and refrigerator off).  So you'd have to use an awful lot more digital attenuation in order to reduce the volume enough that you would be able to hear the difference outside of laboratory conditions.

So with the full -45dB of attenuation and a 120dB dynamic range DAC, you will necessarily be reducing the dynamic range of your music to some extent (even CD music).  But in real-life listening conditions, it's unlikely to be detectable.  For that reason, I wouldn't worry about it; enjoy the music!

[PrinterPrinter]

Thanks Mate,
this stuff is fascinating and really helpful.

So basically don't worry even if I listen to 24/192 with -45dB - I won't really notice the difference?

BTW
does sample rate come into the dynamic range equation?

[mwillems]

Thanks Mate,
this stuff is fascinating and really helpful.

So basically don't worry even if I listen to 24/192 with -45dB - I won't really notice the difference?

You can test for yourself if you have a dB meter or calibrated microphone; you could try setting the volume the same using the digital and analog controls and see what you think.  

But to answer your question, I don't think you're likely to notice any difference with -45dB of digital attenuation with your DAC.

BTW
does sample rate come into the dynamic range equation?

No.  Sample rate determines the maximum frequency that can be represented in the music, not the dynamic range (which is what bitdepth determines).

[mojave]

Sorry, I mixed up loudness with dynamic range.

As far as I know, analog VC lowers the noise floor and digital doesn't
Anyway, what's you opinion about this article?
http://www.esstech.com/PDF/digital-vs-analog-volume-control.pdf

I agree 100% with mwillems.

The ESS article says on slide 5, "As a digital volume control operates on a fixed-width field (ie on that 16 bit number that the DAC receives) it creates noise because the DAC cannot make the fractional part of the number – And this is a large noise!"

Then on slide 15, "When the volume control has access to the additional bits in the DAC data path, there is no numerical loss of accuracy." These comments specifically references the DAC as being able to control the volume at 32 bits instead of 16 bits. Who controls volume digitally at only 16 bits?

I think the article is saying that the noise is due to quantisation errors. If so, then accessing the 32-bit internal data path of the DAC isn't the only solution. You could also access a 64-bit internal data path of the playback software like JRiver does. This will be even more accurate than the 32-bit DAC. In both cases, software or DAC, the final output to DAC chip is 24-bit (slide 14 says the Sabre chip is 24-bit externally). Also, the article doesn't mention that you can basically eliminate the noise caused by quantisation errors by using dither.

By using the argument of the article, JRiver provides a better quality digital volume control than the ESS DAC since it uses a 64-bit data path instead of 32-bit. I realize that the new ESS DAC's have 32-bit output, but 24-bit output with 64-bit volume control is sufficient in my opinion.

[mojave]

No.  Sample rate determines the maximum frequency that can be represented in the music, not the dynamic range (which is what bitdepth determines).

To further clarify, bitdepth only determines how quiet the quietest portions could possible be recorded or how much DSP (including volume control) will affect the audible portion of the media.

It doesn't mean that 24-bit music can play louder than 16-bit or that the dynamic difference of the audible music will even be different. You can convert a media file in JRiver from 24-bit to 16-bit, analyze the new file, and it will still have the same Dynamic Range (R128) and Dynamic Range (DR) values as the 24-bit file.

[mwillems]

To further clarify, bitdepth only determines how quiet the quietest portions could possible be recorded or how much DSP (including volume control) will affect the audible portion of the media.

It doesn't mean that 24-bit music can play louder than 16-bit or that the dynamic difference of the audible music will even be different. You can convert a media file in JRiver from 24-bit to 16-bit, analyze the new file, and it will still have the same Dynamic Range (R128) and Dynamic Range (DR) values as the 24-bit file.

Absolutely; I should've said bitdepth dictates the "theoretical dynamic range" not the actual dynamic range of any given piece of music.

[rudyrednose]

Nice thread!

[dtc]

OT - With PCM the 16/24 bits contain the volume information. How is the initial volume determined for a DSD file? The 0,1s deal with the changes, but where does the initial volume come from? I do not see any reference level in the header.

Thanks for any insight.

[Hendrik]

OT - With PCM the 16/24 bits contain the volume information. How is the initial volume determined for a DSD file? The 0,1s deal with the changes, but where does the initial volume come from? I do not see any reference level in the header.

Its similar to PCM (but different). Instead of the amplitude being encoded using different pulse sizes, the relative density of the pulses corresponds to the amplitude. In short, this means more repeating 1s mean the amplitude goes higher.
You can read a broad overview on this kind of modulation here: http://en.wikipedia.org/wiki/Pulse-density_modulation

[dtc]

Its similar to PCM (but different). Instead of the amplitude being encoded using different pulse sizes, the relative density of the pulses corresponds to the amplitude. In short, this means more repeating 1s mean the amplitude goes higher.
You can read a broad overview on this kind of modulation here: http://en.wikipedia.org/wiki/Pulse-density_modulation

Thanks Hendrik. I think I understand the repeating pattern. What I have never understood is how you get the starting volume. For the first note , how is the  volume determined? How does it know if the first note is at 0 db or at -30 db? Do the first 100 values, for example, set the initial volume relative to some reference? I just cannot find anyplace that discusses that issue. In the Wikipedia article, the sine wave starts at zero. What if it is suppose to start at 1? Does knowing the initial volume require somehow extracting that value from the latter pattern?

Thanks for your insight.

[Hendrik]

There is nothing "to start from". It starts from silence if thats really something you need to define for some reason. But as soon as you start coding audio, you'll have two parts: the frequency and the amplitude. You cannot have amplitude without frequency, and you cannot have frequency without amplitude. So saying "it starts at -30dB" is not something that makes any sense. This goes for both PCM and DSD.

[dtc]

By "starting" I mean the first note, the first sound that is played. With PCM the 16/24 bit values determine that. With DSD is seems that the frequency and order of 1's versus 0's sets that.  I had interpreted the 0/1s to simply correspond to ups and downs in the waveform. It seems the initial run of 0/1s (000000 versus 111111 versus 101010, for example) sets the frequency and volume of the first sound. I am working through understanding that coding better. Thanks for your help.

[6233638]

1) A well-designed analog volume control can always outperform a digital volume control, as they note "if they [the analog volume controls] have a lower noisefloor than the DAC itself."  That is correct, but it is non-trivial given that the noisefloor of the nicer modern DACs are essentially a few dB above the thermal noise of the resistors involved! I've encountered many analog volume controls over the years (even ones made by reputable  manufacturers), that were quite noisey.  Low-noise volume control design is not a dark art, but it's not guaranteed either.  So this point is technically correct in the abstract, but there's no guarantee that the volume knob on any particular piece of kit is going to be lower noise than a well-designed digital volume control (and it might well be quite a bit worse).

If you are looking at a volume control integrated into the DAC itself, I believe you are limited to either Lavry or Burson DACs if you want a really good analog volume control that has the potential to outperform a digital one.
 
Other than that, I guess there are maybe high-end pre-amps which do analog volume control well, but most people are eliminating pre-amps by using DACs that have an internal volume control these days.

2) If you're reproducing 16-bit music on a 24-bit (or 32-bit) DAC, the "performance" of the volume control is entirely irrelevant unless it compromises the 96dB envelope of the original recording.  Put another way, even if you lose 40dB of dynamic range using the digital volume control on a 24-bit DAC, you still have 104dB of dynamic range, which is more than the source material.  Because there is no dynamic range to "lose," the performance of the volume control is a red herring.  With 24-bit music you may be discarding some very quiet bits, but whether bits that are effectively beyond the dynamic range of human hearing (outside of a lab) are important or not is something each of us will have to decide for ourselves.

I think that part of this is also that the ESS DAC can reduce its output voltage - to a point - before switching to entirely digital volume control, which is why it would be better to use the DAC's internal volume control rather than an external one.
 
The internal volume calculations are also performed with 48-bit precision rather than being limited to the 32-bit, or more commonly, 24-bit output from a PC.
 
But in the real-world, most of this does not matter too much. I agree that most of the time a well designed and properly dithered digital volume control will not have a negative effect on audio quality with a 24-bit DAC until you have a significant reduction in volume. When I say significant, I'd say that most people would not start to notice the effects until you are below -60dB.
 
 
If your DAC is capable of ear-splitting or potentially speaker-damaging levels at 100% volume, I would suggest investing in fixed-level passive attenuators.
That way you can keep the digital volume level higher, which increases the signal-to-noise ratio (SNR) and there should not be any of the negative effects that you could have with a variable analog volume control.
 
Of course another option would be to replace the amplifier with one which does not have so much gain - which may bring other improvements, but that is going to be considerably more expensive.
 
With the Chord Hugo specifically, you are probably better off using its internal digital volume control, rather than JRiver's. I'd still leave Volume Leveling enabled, and you probably still want to combine that with passive attenuators so that you can use a higher digital volume level on the DAC though.

[PrinterPrinter]

Thanks guys, this is very interesting indeed.

So just to clarify,
If I listen to a 24Bit file (that MC report to have a '10 DR') in a room that has a noise floor of 30dB - how many bits does it actually use? How much can I attune the volume in MC before losing Dynamic Range?

If the room has a 30dB noise floor, does that mean I can ignore the first 30dB of attenuation (6 bits of DR?) because they are lost in my room anyway?

Thanks for explaining this stuff!

[6233638]

There is no easy answer really.
If you check [Dynamic Range (R128)], that will give you an idea of how much dynamic range a track has, though that is a weighted measurement rather than absolute dynamic range.
[Dynamic Range (DR)] is actually a measurement of crest factor or "Peak-to-Loudness Ratio" not dynamic range. Yes, I know that's confusing.
Noise floor is complicated, because we are able to pick out sounds below the level of the noise floor.
 
As above, get the highest SNR you can by using passive attenuators attached to your DAC's output so that you can increase the digital volume level, use the DAC's volume control rather than Media Center's (though enabling Volume Leveling is fine) and if you're not hearing any hissing from your speakers, you're good.
If it is more convenient to use MC's internal volume control so that you can use JRemote for control, that's probably fine too.
Anything else is worrying about theoretical performance if you aren't hearing any problems from that setup.

[PrinterPrinter]

Thanks mate!

How should I understand the Dynamic Range (R128) stat? If a song has 10 vs 3, what does it actually mean in bits?

[mwillems]

The dynamic range (r128) number is the number of decibels between the 95 percentile for loudness and the 10th percentile for loudness in a song.  The spec is designed that way to gate out sudden very loud sounds, and short very quiet portions (outliers).  So If a song has a 10dB DR(r128) it means that the majority of the sound in the song is within a 10dB range, or about a bit and a half (usually the 10dB closest to the peak, but not always).  

Here's some detailed explanation of how the two dynamic range stats work (the linked post is directly relevant, but there's useful info throughout the thread): http://yabb.jriver.com/interact/index.php?topic=84489.msg577389#msg577389

[PrinterPrinter]

Hello,
I had to revisit this because I may be changing to a DAC with no volume control, just fixed 2v line level - which means I'll have to attenuate by around 50dB - which is 8bits.

Just to clarify, If I'm using JRiver's volume and the DAC is 24 bits - I shoudl be OK with -50dB? At least with red-book material? Is it true that even 16/44 isn't really using all of the range?

Lastly, the DAC I have in mind is the Schiit Yggy - which is actually a 'true 20 bit DAC' - does that mean I have less headroom for volume attenuation than a 'normal 24Bit' DAC?

Thanks!

[mwillems]

Most music has an R128 dynamic range well less than 20dB.  That doesn't mean the music doesn't contain sound quieter than -20dBFS, but it means that 95% of the audio content is between the true peak (usually around 0dBFS) and -20dBFS.  In the case of most modern music, it's actually a much narrower range (a 6dB or 10dB range is not unusual with modern rock music).  

The likelihood that there is meaningful audio content at or near -96dBFS in most redbook recordings is low, and the likelihood that you'd be able to hear it even if it were there is much, much lower.  Even setting aside the extra headroom a 24-bit DAC provides for redbook content, let's imagine the worst case: playing back a redbook recording on a 16-bit DAC, so any attenuation results in some theoretical loss of dynamic range.  

Let me propose an experiment:

1) Play music at your normal listening level for a minute or so.  
2) Now turn down the music -48dB using JRiver's internal volume.  
3) Listen for a minute and note what you hear.

If you digitally attenuated a redbook recording by 8-bits (-48dB), any audio content that you'd lose from the digital attenuation would be at least as quiet as the content you heard in 3).  When I do this experiment, I hear nothing at all in most cases.  If I start from a fairly loud position in 1), I can kind of very faintly hear something in 3).  Armed with that knowledge, you can decide whether that's an issue for you in your setup.

[PrinterPrinter]

Thanks mate,
That's some great advice!

Is there a scientific way (perhaps using JRiver's data) to find out exactly the dynamic range of any particular song? How should the formula look like if I'm using eh Stat for DR 128?

Lastly,
Some claim that we can pick up sounds bellow the noise floor especially timbre and spatial information?

[mwillems]

Lastly,
Some claim that we can pick up sounds bellow the noise floor especially timbre and spatial information?

You can definitely pick up isolated sounds for a certain space below the noise floor of your room especially at certain frequencies (or with certain kinds of distortion), but there are limits.  At a certain point the sound is quiet enough that you simply can't hear it over the noise floor.  And in the vast majority of cases the noise floor of the recording is thirty or more dB below the noise floor of your room which means it should be entirely inaudible, except possibly in headphones with good isolation.

The point of the experiment above is to show how little you can hear with the noise floor in your room when there is verifiable audio content at -48dB.  If you were in an anechoic chamber, you would almost certainly be able to hear the music playing at -48dB. Because you're in a normal room you can hear nothing or very little.  That's largely because most people don't realize that their normal listening level is only about 20 or 30dB or so above the noise floor of their room.

Personally I find the idea that we can "hear things below the noise floor" to be a little bit of a red herring in the context of digital attenuation.  It's technically true but irrelevant in most cases because noise floors are so high and regular listening levels are so low. Loss of dynamic range from digital attenuation is basically a theoretical problem even with truly enormous digital attenuation. 

[PrinterPrinter]

Thanks mate,
I did this little experiment and could hear some very faint details - barely able to follow the music...