Monoprice Monolith Liquid Platinum Headphone Amplifier by Alex Cavalli

Same here. I can’t give up on the Lyr 3 as I feel it still does some things better (a touch cleaner overall and deeper sound stage - the LP is very left-right with the stock tubes, which is my main grip with it.)

 

Just put this together yesterday to give me more range on the volume pot.

 

@Clemmaster and @atomicbob how would you say the Lyr 3 and LP compare in terms of overall tonality? Which is the brighter and more forward of the two? Which sounds more tubbey, warm and/or euphonic? Thanks!

 

Which pot did you go with? A small parts list would be awesome for something super straight forward like this! And maybe a wiring close up/diagram for us who have never done it before

 

A quick update to my initial review:

Tube rolling is definitely easier at this point. They still take some work to remove, but putting tubes in is painless at this point.

I can also confirm what others have said that tube rolling makes a significant difference. I'm currently running a pair of 1972 Amperex A-frames (Holland made), and the sound has become significantly warmer and sweeter. If you find the stock configuration to sound a little dry, then tube rolling can alleviate this. I initially thought this was coming at the expense of staging, but after about 20 hours the sound has opened up and I won't be going back.

 

Shhh...prices on those are now going to skyrocket further....

 

I just tried turning the amps volume all the way up, and using the volume on the attenuator box I built, and notice a big improvement. Extended highs, bigger soundstage, more open sound.
It really has shown me the limitations of the stock, smaller pot. The pot I used in my box is a 50K Alps.
This one to be exact: https://www.ebay.com/itm/1PCS-ALPS-...e=STRK:MEBIDX:IT&_trksid=p2057872.m2749.l2648

 

Now I want a fancy attenuator box ><

 

Try digital volume control instead! Use a software where volume control is part of the upscaler algorithm (eg. HqPlayer, Roon).
Being a software engineer I avoided software v.c. for a long time, because I was well aware of the resolution loss it causes. But eventually I had to realize that it's still much better than analog volume controls. Nowadays I bypass the volume control from my amps. I also sold my preamp. Using digital volume control provides much better SQ.

If you don't want to modify your amp, turn it's the volume control to the maximum. That's where the resistance is a minimum, allowing the least SQ loss.

 

No. As a software engineer, you have no idea what the hell you’re doing. You’re paying for Roon and not cds and records too so you clearly don’t.

I highly advise everyone without anime or cartoon avatars from doing this. One program, os, or driver crash and your tweeters, headphones, and ears could be fried. People with cartoon avatars should let full scale white noise burn in all their gear to stimulate the audio gear economy.

Not with what’s built into audio playback software and without considering the noise floor of your dac. Also pots have channel imbalances at both ends that you’re bringing into your system. There’s no way Cavalli or Monoprice are using a 300 dollar pot like Coleman.

 

I, personally, strongly disagree with this. Even if i used to use volume digital control (because i was lazy to get out of the chair) you can't ignore the risks that this provides WHILE losing resolution/precision, so it's hard for me to see the point other than convenience. Why would you think that it's better than analog volume controls?

 

1, Contact points in the signal path are always a source of problems. Especially if they are based on just one metal touching an another conductive surface, without being squeezed together really hard and having big connecting surfaces.
2, All kind of resistors have their own sound, adding their own noise and distortion.
3, This is what I experienced with my ears

@Psalmanazar:
Using power amp directly with digital volume control is a proven method. It was used long time ago by my Wadia 861, and recently by Metrum Adagio, PS Audio DirectStream, Soekris dac1541, etc. Yes, all of them have embedded computers that might fail, and kill themselves or other devices. Yes, you can accidentally set the volume too high. But it can also happen with analog volume control: you (or your children) can set it accidentally to maximum, causing a surprise when you turn the system on next time to listen Queen's We Will Rock You. Yes, the computer might have bugs, hardware errors, etc. However, the same is true to any other electric device. Life is dangerous
If the amp's gain is so high that it dangers the devices connected to it or the DAC's noise floor raises too high because of high attenuation, that can be solved by soldering a voltage divider as a replacement for the volume pot. It provides a fix attenuation, in other words sets the maximum available volume. It consists of two resistors only, so you can use the best resistors on the market (eg. Z-foils). It doesn't have any moving parts, unreliable contacts, therefore still provides better signal quality than the usual analog volume pot.

...and about my avatar: I understand that it annoys you. Therefore I decided NOT to change it

 

It would take a great deal of convincing for me to ever go back to software volume control, which the above doesn't I'm afraid. The reduction of bit depth that occurs is very much at the expense of SQ, unless you don't like dynamics (I'm making the assumption that you're listening to mainly 16bit files, since all forms of attenuation reduce the bit depth of 24bit files unless you are listening at full range (144db) and blowing out your gear and eardrums).

Running a headphone amp wide open is just maximizing its noise floor. No bueno.

 

Metrum Adagio and Jade use a variable reference voltage for the DAC, not software volume control.

 

No, it doesn't, unless you use only 16 bit processing. Here's why:

Let's assume you have a 24 bit DAC (or PC Digital Output). If you play a 16 bit source, the DAC/Software will simply multiply the received values by 256. The (hexadecimal) result will be like this:
12345678 9ABCDEF1 00000000
After applying the 24 bit digital volume control, the result will be like this:
00123456 789ABCDE 1F000000
(Not perfectly true because the division might change the figures if the divider is not a round power of 2, but it doesn't matter now).
As you can say, the dynamic range doesn't change, because it wasn't there in the beginning. All the original information is there, it's intact. It only had 16 bit of DR, and all of it is preserved.
This is why the better music players use 32 bit processing. Even if your source is 24 bit, the DR will be preserved unless you attenuate more than 48dB (8 bit - which is huge).

However, this is very off topic here, so I will try to remain silent from now on.

@bengo: You're right. I forgot about that. Let's substitute it with the RME ADI-2 in the list of examples.

 

Just a reference point to consider concerning digital volume control.

I'll assume many of you have attended a large scale concert, hopefully one where the venue acoustics, SR system and FOH engineer were all very good. Now look at that control desk (mix console) sending signal to 40 ~ 100K watts of SR. 64 to 128 channels of (*gasp*) digital faders, often grouped into buses with more (*gasp*) digital faders. All of this ends up fed into the final summing bus and out the (*GASP*) digital main bus faders. Let's see some example digital mix console companies:

Allen & Heath
API
Midas
QSC
SoundCraft
Yamaha

You get the picture. SW for those products have fail-safe mechanisms such as watchdog timers, limiters etc. so that run-away events are prevented with exception of catastrophic failure such as lightning hitting mains power near the venue.

Now I've had an RME ADI-2 Pro (about a year now) and RME Fireface UC (for 6 years now) running nearly daily in two labs and have not seen a single crash of the type that would destroy down stream equipment or put ears at risk. I take this seriously as I am responsible for safety of visitors to either lab and answer to OSHA given ability to produce 120 dB SPL 14 Hz to 17 KHz. We log SPL exposure using an ACO Pacific SLARM any time the sound system is operational. In all the years we've had only one event worthy of note, when an apprentice unmuted high level pink noise and fried one of the 18 inch drivers in a D18-E. Limiters did their job but that particular 18 inch driver was not as robust as the rest. Operator error. No SW based volume control crashes.

 

You’re still resampling and dithering. most of these algorithms are colored, especially the older ones used in music players. The better modern DAWs use 64-bit floating point processing are are less colored than say the Foobar2000 volume slider but who listens to music in reaper or logic?

 

Music players and DAWs and most Windows ASIO drivers crash all the time. The better built digital mixers are incredibly reliable. I’d imagine laboratory programs are much more robust too.

Most of use are not using RME or Lynx interfaces with the super reliable drivers into something like a Yamaha O2R though. All of the gear I’ve owned has had less stable ASIO drivers that allowed static noise when Foobar2000 or a DAW crashed. If I was using digital attenuation, a headamp with no pot, and passive speakers, bad things could’ve happened

Edit: for the cost of most stuff, I could just buy a used Yamaha digital console, external multiband eq or plugin, used adat interface, not care about the sound too much, and crank out bangers. O2R and OV1 sound cool in some ways.

 

Can you stop talking for a bit?

I've never had any such issues in years on windows. Maybe the static you hear is just your shitty music?