One thing I hear from objective audiophiles is that amps should strive to be as neutral as possible because you should only hear what the record producer intended for you to hear.
Philosophically I do agree that I should hear what the artist meant for me to hear. However how do you know that the album was recorded while listening to a similarly neutral amp? Recording to a master tape usually doesn't include the amp powering the monitors so whatever they heard through their amp and speakers isn't captured on the tape. In order to really hear what was in the studio, you would need the same amp and the same set to speakers to listen to what they listened to. So if my home setup has a miracle amp and speaker that truly is ruler flat from 1hz-1MHz and 0.00001% THD, I probably still am hearing things differently then the artist because they probably used an average monitor that has -10db at 30Hz.
So how important is it really that an amp be neutral?
Headphone amps aren't any different from speaker amps in principle - what you hear (apart from a bigger version of the input signal) is the power supply's noise coupled through the inadequate PSRR of the electronics.
SE classA operation is a way to minimize the generation of power supply noise by arranging the current flow to be constant to a first order so that any remaining ripple on the supply is the result of the finite output impedance of the follower's loading current source and those of the driving stages. But how significant are these 2nd order effects? This design is an attempt to find out - by reducing them as far as practicable.
The idea is to run SE classA at a much higher voltage than is needed to drive the 'phones (balanced, with 80V supplies giving 144V peak-peak) then step down the output voltage with a custom-wound output transformer. This has the effect of increasing the PSRR of the amp's output stage. I'm not worried overmuch about the PSRR...
I admit I did not give ASUS the benefit of the doubt and seriously consider their Essence STX soundcard as a replacement for my Onkyo SE200-PCI. ASUS make nice motherboards, but unlike Onkyo have no previous expertise in high end audio.
I am happy to report - a bit late in the game, the card came out in 2009 - that they've done a really good job with it and the drivers for Windows 10, technically still in beta, work just fine.
Asus updated the design recently to the STX II. The PCB has been redone, but the only visible change is the BB PCM1792 DAC has been moved towards the top of the card closer to the IV conversion op amps. An second LDO regulator IC, U34, empty on the STX, is now populated. A "TXCO" clock source is added next to the ASUS audio controller IC. The four film caps next to the output IC are replaced with WIMA brand. It's basically identical,...
Current version is 5.60C, last update was 2012 to be compatible with Windows 8. Driver package 5.60C installs without issues on Windows 10 64 bit. The AudioDeck utility installs as well.
However, all is not well:
On installation, Immezio 3D effects are enabled. This locks the sample rate at 48 kHz. 44, 96, and 192 kHz cannot be selected. Deselecting the Immezio 3D effects prompts a reboot, but the Immezio 3D effects remain enabled after rebooting. The card is stuck at 48 kHz. 3D effects (which enables the DSP processing such as Qsounds, EAX, A3D) cannot be shut off.
I see three possible workarounds:
1. Find a command line switch, or edit the installation batch file to disable Immezio 3D on installation...
Posted 2nd October 2015 at 05:07 AM byrjm Updated 8th October 2015 at 04:43 AM byrjm
This is a headphone amplifier with digital inputs, not a DAC with a headphone jack. Though technically given equal board space, the headphone amp, with hot-running single-ended class-A output stage, is surely the centerpiece of the design. (The Asahi Kasei DAC, with MUSES01 for the I-V, is no slouch mind you.)
First impressions. It is large, solid, and very nicely made, but - after seeing the inside - rather simple, spartan even. From the DAC output to the headphone jack is just two op amps and two transistors, the op amps being shared between channels. A third op amp most likely just buffers the analog line output. Apart from the headliner MUSES01 op amp none of the parts are especially expensive, though many were clearly carefully chosen for sound quality - the 2SC5196 for example. The TE7022 USB receiver is a disappointment, as is, to be honest, the single set of power rails and the use of dual op amps shared between channels.
Posted 26th September 2015 at 11:49 AM byrjm Updated 9th October 2015 at 04:24 AM byrjm
My Onkyo soundcard drivers stopped working when I upgraded to Windows 10. Onkyo says they have no plans to release a patch, so I'm left with no high quality audio solution for my computer. Since I already have a good headphone amplifier, what I'm mainly looking for is a high quality line level analog output.
One options is another soundcard, the ASUS Xonar STX being the obvious choice. I dunno, it doesn't grab me.
I was thinking with going with an external box this time, connected via USB. As this opens up about a zillion options, I'm going to limit things to,
Respected audio brands with a solid reputation for digital audio.
Small enough to be placed on top of my computer case.
Posted 25th September 2015 at 01:56 PM byabraxalito Updated 26th September 2015 at 01:35 AM byabraxalito
Since I figured out the reason for needing all those caps in my earlier DAC designs was all brought on by using passive I/V, I'm now a total convert of active I/V in order to do away with the sheer bulk.
Having tried single transistor I/V and loved it, I found there was still some improvement to be gained by biassing the common-base transistor with additional current sources to reduce its input impedance. Since getting down to the region of 1ohm would require some 25mA of bias which isn't well suited to portable applications I decided to have a go at using feedback to obtain the impedance I'm seeking.
I'm not using an off-the-peg CFB amp because they still turn out to be fairly power supply quality susceptible (subjectively speaking) so here's a design I hope that greatly reduces the supply impedance requirements so that it can be used in a portable player.
The picture shows the second prototype I/V stage, coupled to a 6th order Chebyshev anti-imaging...
Right down at the bottom of the page the last filter he shows the schematic of is a 9th order Chebyshev, 1dB ripple, with a corner frequency of 1kHz. A textbook frequency response plot is obtained using LTC6241s. I latched on to this and tried changing the corner frequency to 18kHz, wondering if I could use such a design for an anti-imaging filter for my DACs. So I divided all the capacitor values by 18 and ran the sim. Disaster! The frequency response I obtained is below - a 7dB spike at 17kHz.
The problem seems to be inadequate Q - high order filters are composed of sections which increase in Q (more positive feedback) and the chosen opamps aren't fast enough (18MHz GBW). I went to a faster opamp for the highest Q stage which brought about some improvement...