Can source components and better quality signal path make a difference ?

[Warpdrv]

Ok I wanted to take components of the other threads discussion points so as to not derail any further....

I'm not an engineer and haven't been to the moon, but as G-EV noted I completely implied the word slight - and I meant exactly that... on that note I would love to see the different measurements between a tube amp, class d and a solid state amp just to see what the potential differences are in perhaps a square wave measurement or something along those lines - just to see what the measured differences are and how that translates to the signal put out to the speakers... Being completely open minded and honest here, no mal intent towards the discussion.. I would truly like to see the differences in the measurement - something I have never seen on any forum ever - I am truly very interested at this point...

FYI - I'm in absolutely NO WAY a snake oil purchaser - no way I ever buy into that idea ever in my life. But I can only know what I hear in the differences.


I find that the more you invest in the utmost in revealing speakers it truly starts to translate up the river of electronics involved in the chain... Some people will also state unequivocally that electronics are all the same, and from my experience it couldn't be farther from the truth...

I can feed my Pioneer receiver a 2 ch HDMI FLAC signal and toss it out to my amps to my Salk M7's (totally lifeless flat collapsed soundstage) - or I can extract that same 2 ch audio from the HDMI signal toss it out to my Parasound 2100 with HT bypass and it portrays a completely different soundstage with the push of a button, level matched with meter. Throw a really good quality DAC in the mix and its a whole new presentation... I in all honesty am not sure how you would measure that - but there is no question from the countless people that I have had listen to the same thing - it was world of difference.

Seriously push of a button -
I'd be more then happy to A/B that scenario for anyone that is interested.

It's only conjecture, but it sounds to me like your amps are low sensitivity while your receiver has weak "throw away" preouts. I honestly don't have a clue what exactly is happening, but it sounds like this is one of those case of operating outside its limitations. I'm not convinced it's just "2 channel preamp" magic. Obviously if there's a major audible difference, it's something.

the other possibility, is that your receiver is doing something funky to the phase relationship between the speakers, warping your soundstage. Did you set speaker trims and distances manually by any chance?

I think at the end of the day, any comparision done at home is always subject to countless unknown, unidentified sources of error.

I'm not gonna lie, this one's a pretty dubious claim, unless the DAC is the audiophile sort that has built in EQ to appeal to people.

Its pretty obvious its not my first rodeo in setup, I run no room correction on my receiver, all manual setup, phase - distance - trims... I only run EQ on my subs - there is no room correction in play with this system for either the Pioneer or Parasound.


Please know that I'm not one to spin this into some audiophoolery realm. I'm very much intrigued by the clearly audible sonic differences here... I'm not speaking of rainbow floating unicorns or any goofy Cr@P, just some honest experiments our small group of local guys have messed with, and found improvement.

This is not the first instance where I / we have experimented with this same setup. The WI GTG boys have all implemented this HT bypass scenario into our mix. After experimenting through A/B tests on each of our systems over and over again - no question how much the differences are. The other guys all have different brand receivers, yamaha, pioneer, onkyo, and we are all very intricate in setup. The sources are all digital streaming FLAC files.

With the HT Bypass - you send the preouts of the receiver through the Parasound and output to the amp. Push the button on the Parasound and it takes over the Signal path. Its an instantaneous switch back and forth - always level matched to be equal....

The Atlona HDMI deembedder has its own built in DAC (which would then send the analog signal via 1/8" jack to RCA into the Parasound input) - I can't speak very highly of its quality in conversion, sound was edgy... Utilizing the Optical out of the Atlona, bypassing Atlona DAC and sending out digital signal to a couple different higher end DACs - the SQ and soundstage was again noticeably better... I'd assume a far better, far cleaner build quality to the components in the signal path.... all RCA cables are just good quality monoprice...


The investment was minimal -
Parasound $525
Atlona $200
External Dacs were not mine $0

So here is where I wonder = everyone states that there is no difference in electronics. Yet its undeniable to multiple people in the same room in each one of our different systems A/Bing in an identical manor that there is a change in presentation.... How does one measure such differences.... I'm not saying I'm capable of these measurements (no experience with an oscilloscope or other tools of this nature) , but I would like to understand what is happening here and the reasons why....

Can one honestly say that there is no difference in electronics... There is science in building electronics Yes / No ?
Why is it so far fetched that there can't a better quality piece of electronics be engineered for better handling of signal and improved SQ ?

Warp

 

[PENG]

Its pretty obvious its not my first rodeo in setup, I run no room correction on my receiver, all manual setup, phase - distance - trims... I only run EQ on my subs - there is no room correction in play with this system for either the Pioneer or Parasound.

May be there is something about that particular Pioneer. Have you tried another mid range AVR, it could be just another Pioneer or Yamaha, or Denon etc.?

Please know that I'm not one to spin this into some audiophoolery realm. I'm very much intrigued by the clearly audible sonic differences here... I'm not speaking of rainbow floating unicorns or any goofy Cr@P, just some honest experiments our small group of local guys have messed with, and found improvement.

I hear what you are saying and I am not sure why you and your group share such experience. In my case we (me and a few friends) had similar experience in earlier years and then we found out if we forced ourselves to not think about prices and hearsays (reviews) and focussed on doing our best in comparing different gear properly, we realized we were not hearing the kind of differences we thought we were. And that's before we heard about those double blind tests with rewards. I am not sure why different groups of people have different experience in this regard. It is one of those things that is strange, but true.

The other guys all have different brand receivers, yamaha, pioneer, onkyo, and we are all very intricate in setup. The sources are all digital streaming FLAC files.

That almost answers my first question, no help though.

So here is where I wonder = everyone states that there is no difference in electronics. Yet its undeniable to multiple people in the same room in each one of our different systems A/Bing in an identical manor that there is a change in presentation.... How does one measure such differences.... I'm not saying I'm capable of these measurements (no experience with an oscilloscope or other tools of this nature) , but I would like to understand what is happening here and the reasons why....

Can one honestly say that there is no difference in electronics... There is science in building electronics Yes / No ?
Why is it so far fetched that there can't a better quality piece of electronics be engineered for better handling of signal and improved SQ ?

I have seen square wave comparisons on magazines, Stereophile magazine being one of them. Obviously a spectrum analyzer can analyze the differences between them in quantifiable ways. If there are conclusive reasons for those "slight" differences to be audible there would have been publications about it but I have seen none. A square, or near square waveform is simply a series of sine waves of different frequencies (harmonics). It is apparently known that even (2nd,4th, 6th....) harmonics give us the so call "warm" sound characteristics and that could explain why tube amps tend to give us that impression. Other than that, most mid range amps should produce THD and IMD well below the level that is discernible.

You know there are also many who believe in breaking in their electronics by playing them continuously for may hours while perhaps even more people would swear they heard no difference between one came straight out of the box and one that had many hours on it. I know this is not relevant to your post, but it is another example of things we really don't understand much about. I often thought the topic we are discussing here would be an excellent university research subject for their PhD students in various disciplines.

 

[highfigh]

I have seen square wave comparisons on magazines, Stereophile magazine being one of them. Obviously a spectrum analyzer can analyze the differences between them in quantifiable ways. If there are conclusive reasons for those "slight" differences to be audible there would have been publications about it but I have seen none. A square, or near square waveform is simply a series of sine waves of different frequencies (harmonics). It is apparently known that even (2nd,4th, 6th....) harmonics give us the so call "warm" sound characteristics and that could explain why tube amps tend to give us that impression. Other than that, most mid range amps should produce THD and IMD well below the level that is discernible.

An imperfect square wave may be "simply a series of sine waves of different frequencies (harmonics)" but the square wave used for testing audio equipment is a true square wave. The trick is in designing amplifiers and other circuits that can reproduce it without overshoot and ringing. Some amplifiers can't even produce a clean sine wave without excessive harmonics ('excessive' being relative).

A spectrum analyzer, as you refer to it- is this when testing for harmonic distortion? If so, it's useful but less so when comparing a harmonic-rich signal unless the distortion is extreme or the input waveform can be overlaid on the output, with some kind of marker so it would be known that the two are in synch, rather than taking part of one wave and comparing it with part of another.

Part of the problem when comparing equipment is that we need to ask "Which kind of signal is the most useful when testing for distortion?", "How much distortion of the waveform is too much?" and "Is distortion really the most important quality to test for, or should we be looking for something else?". If we use sine wave, someone will come along and say that square wave is a better way to test. Then, someone else "After all, music is made up of one or more instruments playing one or more notes at the same time". Regardless of what is used, we need to find out "What has been distorted, how much has it been distorted and how was it distorted?". Is it only happening with waveforms that approach square wave? Is it a phase error? Is it a dynamic range issue? Is it a combination?

 

[PENG]

An imperfect square wave may be "simply a series of sine waves of different frequencies (harmonics)" but the square wave used for testing audio equipment is a true square wave. The trick is in designing amplifiers and other circuits that can reproduce it without overshoot and ringing. Some amplifiers can't even produce a clean sine wave without excessive harmonics ('excessive' being relative).

A spectrum analyzer, as you refer to it- is this when testing for harmonic distortion? If so, it's useful but less so when comparing a harmonic-rich signal unless the distortion is extreme or the input waveform can be overlaid on the output, with some kind of marker so it would be known that the two are in synch, rather than taking part of one wave and comparing it with part of another.

Part of the problem when comparing equipment is that we need to ask "Which kind of signal is the most useful when testing for distortion?", "How much distortion of the waveform is too much?" and "Is distortion really the most important quality to test for, or should we be looking for something else?". If we use sine wave, someone will come along and say that square wave is a better way to test. Then, someone else "After all, music is made up of one or more instruments playing one or more notes at the same time". Regardless of what is used, we need to find out "What has been distorted, how much has it been distorted and how was it distorted?". Is it only happening with waveforms that approach square wave? Is it a phase error? Is it a dynamic range issue? Is it a combination?

If there is no harmonic distortion the harmonics reproduced would be exactly the same as the original. A perfectly square waveform can stil be resolved into a series of sinuisoidal waveforms of different frequencies. You can research the topic of Fourier series on this. No need to go to the library or university text books either.

Square wave - Wikipedia, the free encyclopedia
Fourier series - Wikipedia, the free encyclopedia

Be carefull when you see the word "approximate", it may start that way but like Calculus, the end result is exact, and yes, accurate enough to get us to the moon, the cell phone and other communication technologies etc.

 

[ski2xblack]

I like these kind of threads, so I'll kick in.

I too believe that there may indeed be differences in source components and pres and amps, but that there are no unexplainable differences. They can be accounted for somehow, and the explanation had better be reductive, believable, and convincing.

That being said, I battle medical woo in meat-space, and when I get home and kick back, I'm a bit more forgiving of what most here would consider woo in audio-land. See avatar for an example of what I use in one of my systems.

Despite that I'm willing to play with high output impedance tube amps, I don't believe in magic wire, caps, maple plinths, pyramids or super sources. My boring digital front end consists of a collection of players that span the decades, from an old Sony ES to a new BD player. Level matched, and in spite of completely different DACs and output stages, they are indistinguishable. When those players break, I'll move on and get a music server of some sort. I expect what comes out to also be indistinguishable, if not better, than what I get now. And it will be hooked up with cheap interconnects and speaker cable. My old cheap homebrew 12g cables must be thoroughly "burned in" by now, they're extra chocolaty.

I would love to see the different measurements between a tube amp, class d and a solid state amp...

This is old news, but it has been reinforced to me with some inspiration from Bob Carver and folks here at Audioholics. A while back, someone posted a thread about one of Bob's recent offerings on eBay. On the listing, Bob was answering questions, and the discussion was quite interesting.

Regarding tube vs ss sound, Bob gave the expected answer, that the big difference was the output impedance and resulting misbehavior, but he went a bit further than what I had heard before. He postulated that in addition to the back emf from undamped driver excursion, the low damping actually allows the speakers to pick up room response like microphones, which, since it isn't damped out, instantly becomes part of the signal. The effect would be more noticable as output impedance went up, so typical PP tubes would have a little of the effect, and high output impedance SETs would have even more.

So I had to check it out, since I have SETs in a bedroom system to experiment with. Their output impedance is about 3.2 ohms.

Such behavior is actually happening in my room. I hooked a multimeter to one channel, completely disconnected from the source but with amps on and gains maxed. While cranking the other side, sure enough, I was getting a signal in the unconnected channel, way down in the low milliwatt range, but there. Volume dependant, of course. It goes a long way to explain the 'bloom' effect of using high output impedance tube amps, aside from the low order but considerable amount of distortion they produce. It's a more natural sounding reverb than I could get with a processor, and does quite pleasant things to the soundstage. It also explains why the effect is exclusive amp/speaker connection, and can't be exactly duplicated with processors.

I don't see any reason why tube circuits handling low level signals would perform any better or worse than their ss equivalents, unless by specific intent. I've tried a few, and just don't see the point, unless you're addicted to tubes for their glow. I guess I like my signal clean up to the point that the SETs/speakers bastardize it, and not before.

This particular corner of audio-geekery is a dead-end, I'll admit. But that doesn't mean that my weird franken-system that breaks all the rules doesn't get it's share of "Wow, that's amazing!" type responses. And that's all that matters, if it connects the listener to the music in an emotional, engaging way. Otherwise it's just noise.

Regarding Warp and his buddies' experience, I'm kind of with Peng and the others, in that with more careful control some or all of the perceived differences may indeed be illusory, but they are explainable somehow. I wouldn't be so quick to discount what they're hearing; the ear is the final judge, after all, even if it is influenced by subconscious biases. There is something aesthetically appealing about a short, simple signal path, even if a more complex one is sonically indistinguishable, and that alone will affect the listeners perception.

 

[mtrycrafts]

Ok I wanted to take components of the other threads discussion points so as to not derail any further....





Its pretty obvious its not my first rodeo in setup, I run no room correction on my receiver, all manual setup, phase - distance - trims... I only run EQ on my subs - there is no room correction in play with this system for either the Pioneer or Parasound.


Please know that I'm not one to spin this into some audiophoolery realm. I'm very much intrigued by the clearly audible sonic differences here... I'm not speaking of rainbow floating unicorns or any goofy Cr@P, just some honest experiments our small group of local guys have messed with, and found improvement.

This is not the first instance where I / we have experimented with this same setup. The WI GTG boys have all implemented this HT bypass scenario into our mix. After experimenting through A/B tests on each of our systems over and over again - no question how much the differences are. The other guys all have different brand receivers, yamaha, pioneer, onkyo, and we are all very intricate in setup. The sources are all digital streaming FLAC files.

With the HT Bypass - you send the preouts of the receiver through the Parasound and output to the amp. Push the button on the Parasound and it takes over the Signal path. Its an instantaneous switch back and forth - always level matched to be equal....

The Atlona HDMI deembedder has its own built in DAC (which would then send the analog signal via 1/8" jack to RCA into the Parasound input) - I can't speak very highly of its quality in conversion, sound was edgy... Utilizing the Optical out of the Atlona, bypassing Atlona DAC and sending out digital signal to a couple different higher end DACs - the SQ and soundstage was again noticeably better... I'd assume a far better, far cleaner build quality to the components in the signal path.... all RCA cables are just good quality monoprice...


The investment was minimal -
Parasound $525
Atlona $200
External Dacs were not mine $0

So here is where I wonder = everyone states that there is no difference in electronics. Yet its undeniable to multiple people in the same room in each one of our different systems A/Bing in an identical manor that there is a change in presentation.... How does one measure such differences.... I'm not saying I'm capable of these measurements (no experience with an oscilloscope or other tools of this nature) , but I would like to understand what is happening here and the reasons why....

Can one honestly say that there is no difference in electronics... There is science in building electronics Yes / No ?
Why is it so far fetched that there can't a better quality piece of electronics be engineered for better handling of signal and improved SQ ?

Warp

Hard to say what or where the cause is as you are doing different things. What does that HT bypass doing? Does the Pioneer have a similar feature?
Don't know about the FLAC file and how that may be treated differently.

What happens with a CD music? How about a test tone either single tones, that full band noise kind? How about a BD movie? Are these other sources being affected by the components or just the FLAC? Lots of unknowns that needs to be nailed down for a reliable outcome.

 

[3db]

The mind is a very scarey super computer in its ability to discern the senses. Warp, I'm not saying you aren't hearing things differently but it may be your perception thats hearing and not your ears. The best analogy I can come up with is trying to see things in the dark. Our minds plays all sorts of tricks on us based on what we're thinking/feeling and we come up with a list of possibilities of what that object is. Its not until we get close enoug to the object that we can truly figure out what that object really is. Removing the sight biases from your A/B testing and removing the knowlede of teh equipment being tested would go along way and removing some of the preconceived biases that we all have.

 

[Warpdrv]

Removing the sight biases from your A/B testing and removing the knowlede of teh equipment being tested would go along way and removing some of the preconceived biases that we all have.

I agree 100% - but remember that all my equipment is located in the basement aside from the speakers - so the sight biases are removed...

The person listening knows I'm making a change, but cannot see any changes being made to equipment - as its all done at the push of a button on my remote.... It has been very easy for those to differentiate the two different presentations on the subjects that I have had listen for the difference.

Level matched before the trial at the LP. Then its just a matter of pressing play.

 

[GranteedEV]

i would like to see a receiver that has actually powerful measured pre-outs, like the marantz sr6004 (7vRMS) and how it compares to the parasound. I suspect all the other receivers tested had weak preouts.

 

[Warpdrv]

I'm not going to argue with that - never measured the pre-outs of the Pioneer, but had no issues driving my Behringer EP2500 for my LMS, as many have had with other receivers in the same position.

I wonder it that is something that can and will have a difference in the SQ / SoundStage.... its a noticeably bolder presentation... But again - they are level matched, so that IMO would remove any notion that there not at the same volume where people would cite the higher SPL unit will always win hands down in pref...

Again - you guys all know my position that I'm not here slamming my fists on the table saying "YOU LISTEN TO WHAT I SAY DANGIT - ITS DIFFERENT"...... Just not my style...

 

[GranteedEV]

I'm not going to argue with that - never measured the pre-outs of the Pioneer, but had no issues driving my Behringer EP2500 for my LMS, as many have had with other receivers in the same position.

ep2500s are pretty sensitive amps and sub outputs are usually good. main outputs on the other hand often get the 'meh' treatment, and hi fi amps are often low sensitivity to have the lowest noise floor. the levels are probably matched at 70-80db but if the preouts are soft clipping or something... that's going to squash some life out of stuff.

Even the $2000 Yamaha A3000 'only' measured with around 2.8v pre outs in the measurements. The sr6004 doubles that and retailed for less and the refurbs are like 400 bucks. man am i shill for marantz or what.?

TLS Guy measured his AV8003 processor with 14v preouts.

 

[mtrycrafts]

....

Again - you guys all know my position that I'm not here slamming my fists on the table saying "YOU LISTEN TO WHAT I SAY DANGIT - ITS DIFFERENT"...... Just not my style...

I fully understand you and am sure others are on the same page, you want to know what is going on. Perhaps you are too close to the setup to spot the problem.

 

[highfigh]

If there is no harmonic distortion the harmonics reproduced would be exactly the same as the original. A perfectly square waveform can stil be resolved into a series of sinuisoidal waveforms of different frequencies. You can research the topic of Fourier series on this. No need to go to the library or university text books either.

Square wave - Wikipedia, the free encyclopedia
Fourier series - Wikipedia, the free encyclopedia

Be carefull when you see the word "approximate", it may start that way but like Calculus, the end result is exact, and yes, accurate enough to get us to the moon, the cell phone and other communication technologies etc.

Square wave, especially at high frequencies, is a great way to see how fast a circuit can react to the input signal but that's about all, unless the source material will be synthesized audio without processing. A spectrum analyzer is a great way to see the level of THD/IM distortion. In theory, these should tell us that the signal is being processed and/or amplified faithfully, but apparently, there's more to it than just a few measurements.

As long as sine waves are the signal being processed and amplified, THD is a great spec but when complex waveforms and multiple frequencies are coming through at the same time, IM, phase shift and other issues come up. I do think that we need to standardize amp testing results- in the '70s, the FTC required that the THD spec come with "both channels driven, from xxHx-xx(x)KHz, ±xdB", not the useless @1KHz, each channel crap. I have an integrated amp that's rated in The Vintage Knob as "100W/ch, 5Hz-35KHz power band, .03%THD/.015%IMD, 8 Ohm load" but I remember my owner's manual's specs being better.

Ya know, if my calc teacher had explained some of the practical uses for his instructions to "integrate the area under the curve", I would have done much better. It's all his fault!!!!!

 

[highfigh]

I'm not going to argue with that - never measured the pre-outs of the Pioneer, but had no issues driving my Behringer EP2500 for my LMS, as many have had with other receivers in the same position.

I wonder it that is something that can and will have a difference in the SQ / SoundStage.... its a noticeably bolder presentation... But again - they are level matched, so that IMO would remove any notion that there not at the same volume where people would cite the higher SPL unit will always win hands down in pref...

Again - you guys all know my position that I'm not here slamming my fists on the table saying "YOU LISTEN TO WHAT I SAY DANGIT - ITS DIFFERENT"...... Just not my style...

Are you using an SPL meter or RTA software? I think you may see a slight difference in the response of the equipment with the "bolder presentation".

 

[Warpdrv]

SPL meter....

I have all the equipment for REW with the EMM-6 calibrated mic (not the expensive calibration), just got to get it configured on my computer.

Also, getting REW hooked up to this system is not a walk in the park with all the equipment located in the basement.

 

[TLS Guy]

Can one honestly say that there is no difference in electronics... There is science in building electronics Yes / No ?
Why is it so far fetched that there can't a better quality piece of electronics be engineered for better handling of signal and improved SQ ?

Warp

Of course there is a difference in electronics. It is not that difficult to understand.

A big part of the problem in understanding this, is that manufacturers now keep their circuits a "state secret". If they were available you could get a lot of answers just looking at the circuits.

As you know I hammer away about head room. Why? Music has an enormous dynamic range.

Line voltage is nominally 1 volt. However a line input or output that clips close to a volt is not useful. Almost all reviews are done not exceeding manufacturers spec.

Now a chip can not output more volts than the supply voltage.

A lousy unit quite likely will power the preamp section with a five volt supply and I have seen as low as 2.5 volts! Now a unit powering the preamp with a line voltage of 2.5 volts and one with +/- 15 volts, will test the same, but they won't sound anywhere close the same on music.

The low voltage one will be clipping all the time on most music and collapse the stereo image as your describe.

Now the higher voltage unit requires more expensive and physically larger caps than the lower voltage one. The cost to build will be markedly different. In most receivers the higher voltage stages I would bet are nowhere to be found.

Now I have pointed out time and time again, that a receiver with small pairs of output transistors will be no match for an amp with large triple devices.

So why do a lot of people not hear a difference? Speakers, speakers and speakers. If the speakers are poorly designed (most), then there will be massive thermal compression due to V/C heating. So the thermal compression will tend to make the poor quality unit sound like the better one.

I'm still in England and spent a delightful morning with Billy Woodman in their factory deep in the Cotswolds, between Gloucester and Cirencester. They have chickens and a large vegetable patch to keep the employees in eggs and vegetables.

Anyhow Billy Woodman was trained as a transducer engineer and I regard him as probably the worlds foremost transducer engineer.

He stressed the importance of having lots of amp power and transducers that really minimize thermal compression. When I get back I will post more about the extraordinary lengths you have to go to with a speaker driver to achieve this.

Just to wet the appetite I saw parts of a center speaker being built for an auditorium at Stamford University that will weigh in at just over two tons!

So yes, electronics makes a huge difference in numerous ways. However you have to have the speakers to hear it.

 

[PENG]

Ya know, if my calc teacher had explained some of the practical uses for his instructions to "integrate the area under the curve", I would have done much better. It's all his fault!!!!!

I hear you. It wasn't until many years after graduation when I had to repeat a course in communication that I began to understand the practical use of Fourier transforms, let known Laplace, Z and other mathematical analysis/transforms.

Fourier not only provided the means to analyze waveforms effectively, it also allow us to go back and forth between the time domain and the frequency domain. One can easily understand how without Fourier we would not have the telecommunication technologies we all enjoy and take advantage of in such a big way. Our beloved Audyssey also involves the use of Fourier theory.

 

[PENG]

So Warp, I am sure TLSGuy knows way more than me in the design/build of loudspeakers. However, the point he made about the need to have lots of power was not much different than the point I tried to make previously when I said I wouldn't give a crap (again just borrowing words not originated by me) about some of those claims of difference heard. That is, you will hear a difference, but not under any conditions. Examples of those conditions are, as TLSG stated or alluded to, speakers, speakers..., music that has the kind of dynamic contents he referred to etc.

I would also add that the source media has major effects too, i.e. the quality of the specific SACD, DVDA, CD, BR, Vinyl as well as the SPL we are listening to. If you listen to Diana Krall jazz at average SPL of 70 dB in a 20X15 room sitting from 12 ft away your decent AVR such as a 4311, A3000 is going to be as good as a XPA-5 because the music will probably never peak to anything near 85 dB. I have a few Telarc CDs that sound way better than some of my SACDs, so the media format alone is not necessary the deciding factor either.

By the way, I do remember you have very nice speakers so at least that is one of the possible reasons in your particular case.