I like your term ‘bottleneck analysis’. We should use this term more often to reach realistic common sense ground.
Some things that come to mind:
Cables – by the time we insert our majik cables in our home audio or HT system, the recorded signal has already been transmitted through up to hundreds of feet of non-majik cable.
Opamps – By the time we use an opamp or other kind of transistor circuit, a recorded signal has already been through up to hundreds of transistors in its path.
Passive components – Too large to fully comment but I’ll make a couple of comments. I see carbon composition resistors are in vogue in audiophoolery circles
. I still have a whole box of ‘Wonder Caps’ They make you
wonder.
I wonder how many of these parts are in the recording chain...
Audiophile fickleness – We get easily bored and need to move on to the next toy. Occasionally we stop and listen to the music. I never get tired of Bach’s Solo Cello Suits. My current favorite is YO-YO MA, The Bach Project, Cello Suits J.S. Bach, From the Odeon of Herodes Atticus Athens. DTS-HD MA 5.1 recording. Complete with traffic and airplane noise. Signal to noise is probably about 40dB at best. When I get tired of every other piece of music, I put on a version of Bach Solo Cello Suits – or nothing at all.
Limits of knowledge – Most people can’t juggle 2 balls let alone all the ones in a multidisciplinary field.
Industry mysticism – the last thing the industry wants to do is fix real problems. It’s an IC – Industrial Complex. It’s sort of like the MIC – Medical Industrial Complex – that manages diseases rather than seeks cures. The major drug that the EIC – Entertainment Industrial Complex – dispenses is ‘Havitol’. Just do everything the industry says and you can havitol.
IMO this is a real winner rated post! It explains a lot, though never totally I supposed, why "all amps sound the same" and "all amps don't sound the same" are both a matter of "facts".
As mentioned (or alluded to), by the time our ears and brains get to even start working, perceiving, the signal, since created in the recording and mastering process, has gone through so many electronic parts, connectors, cables, wires, solder joints etc etc etc...., so from one of the numerous online dictionaries:
when trying to do a bottleneck analysis on the audio signal chain, from the recording mics through the loudspeakers or headphones, then to the ears and brains, one must keep the following in mind, unless one is able to ignore what I call "logic":
- A Chain is As Strong As The Weakest Link
- A Chain is No Stronger Than Its Weakest Link
- A Chain is As Strong As Its Weakest Link
- You’re Only As Strong As Your Weakest Link
Now apply to the bs claims of HDAMs, let me repeat my logic:
a) Marantz AVR and AVPs: HDAMs cannot improve slew rate (see definition on Sweetwater website):
Slew rate is the ability of a piece of audio equipment to reproduce fast changes in amplitude. Measured in volts per microsecond, this spec is most commonly associated with amplifiers, but in fact applies to most types of gear. In amplifiers, a low slew rate “softens” the attack of a signal, “smearing” the transients and sounding “mushy.” Since high frequencies change in amplitude the fastest, this is where slew rate is most critical. An amp with a higher slew rate will sound “tighter” and more dynamic to our ears.
So if the op amp before and after the HDAMs are the same as those used in the audio signal chain of the corresponding AVRs, as well as Marantz own AVRs but the low profile series, then even if the slew rate of the HDAMs are "infinitely" fast, it will not mater become it ends up going through the same opamps that have much slower slew rate (assuming Marantz if right about the much faster slew rate of the HDAM opamps (yes HDAMs are opamps, just discrete, not IC type)
-b) Marantz integrated amps: HDAMs in those will more likely realize Marantz claim on faster slew rate, because they are used in quite a few more stages, vs the AVRs/AVPs, where they are only used in the stage just prior to the final opamp buffer (again, yes, I saw it in the schematics/block diagrams of the SR6014 that I purchased when it was available).
c) as PaulBe pointed out, HDAMs, even just for one single stage in the AVRs/AVPs, will likely have different output impedance than units without HDAMs.
d) HDAMs, should in theory, if Marantz had chosen to, improve the pre out voltage SINAD at higher voltage when driving power amps that has relatively lower input impedance, that is, they could be more effective in terms of "buffering".
Overall, HDAMs unlikely has audible effects, that's probably why Marantz, since the AV10, and the Cinema 30 has included the selectable dac reconstruction filter, so that the user could select the default or the optional filter as follow (Marantz website):
Marantz models such as the AV10, CINEMA 30, MODEL M1, and MODEL M4 offer a “DAC Filter” option within Audio settings. These DAC Filters adjust the roll-off characteristics of the audio signal.
- Filter 1 (Default): Recommended setting to enjoy the Marantz sound.
- This filter has short delay slow roll-off characteristics.
- Filter 2: Recommended setting for bench test measurements.
- This filter has sharp roll-off characteristics.
This particular claim while just a claim because "audible" or not is debatable and it certainly depends on the individual's hearing ability, or discernibility in the high frequency range as well as being subjective in nature, it is however, at least 100% logical, even when bottleneck analysis is correctly applied.
Bottom line,
@TLS Guy has already confirmed the AV10 has superior surround sound performance, and is very audible to him, I hope he will try "Filter 1" and "Filter 2" and report back on whether he could easily hear a difference and then which one he prefers.
If I remember right, Gene said he could, or he thought he could? hear a difference and he prefers Filter 2 (NOT 100% SURE IF I INTERPRET THAT CORRECTLY FROM HIS TALK), that means he would have been just as happier with the Denon (A1H, A10H) sound too.
Finally,
@ryanosaur , can you please also try Filter 1 and Filter 2 and gives us some feedback, thank in advance.
NOTE: Gene told us his preference between Filter 1 and 2 but he used the term that he prefers the one with the "wider bandwidth" so while I am somewhat confident he meant filter 2, there is an outside chance he might mean filter 1 because filter 1 will result in a roll off of about 2 dB by 20 kHz and about 2.2 dB at around 22 kHz, filter 2 will drop more at 22 kHz as it would act more like a brick wall at that point (ref: Nyquist criteria)
So at up to 44.2 kHz sampling, one can consider either filter as having wider bandwidth, I lean on believing Gene meant he prefers filter 2 because a) at 88.1, 96 kHz or above sampling, filter 2 definitely and clearly has wider bandwidth, and Gene mentioned it looked better on the test bench, since most bench test measure 20-20 kHz, and on that one, Filter 2's FR looks like a straight line vs Filer 1's roll off from about 12 kHz, so Gene most like meant Filter 2 look better on measurements. Marantz basically has the same narrative on that too:
Regardless, if
@gene happens to notice my post,
please clarify that your preference is in fact filter 2, or 1 (edit again: just realize in his review on this website he did make clear the one he preferred was the Filter 2), and thank you again for such in depth review plus a even more in depth review in the Youtube follow up video.
I would however, urge you to do another one when you have tried out DLBC, as I am quite sure you will get even better, smoother bass response with DLBC, with some minor tweaking. I am saying this based on my own extensive trial comparing Audyssey, ARC G and DLBC.
