Category Archives: Audio


HRTF is Head Related Transfer Function. It describes how you perceive sound. Every person perceives sound differently because the individual shape of your head, ears, nasal & mouth cavity, etc. all affect how the sound reaches your ears. In short: different people listening to the same thing, hear it differently.

What most HRTFs have in common is the range from 2 – 5 kHz is amplified by 15 dB or more. The ear’s resonance is typically +17 dB at 2.7 kHz. That is a huge non-linearity. Here is a typical HRTF curve from Tyll Herstens at Inner Fidelity.

Another way to think about this: Suppose you’re standing at the seashore listening to waves crashing on the beach. That sound is similar to white noise: it has roughly equal energy across a wide frequency range. The sound you actually perceive, however, is 10 – 20 dB louder in the 2 -5 kHz range because those frequencies were amplified (or frequencies outside that range attenuated) by your head, ears, ear canals before it hit your eardrums.

You can easily test how the size & shape of your head & ears affects sound. While listening to music on speakers, gently push your ears forward or open your mouth really wide. The sound changes. And that only gives a small taste of what the real differences are – imagine how much more different it might be if you could change the size & shape of your head, ears, etc.! That different sound you hear would be what another person hears normally.

The astute reader will wonder – if this variation is due to individual variance in body size & shape, how can it be measured? The answer is simple. Take 2 tiny microphones small enough to fit inside your ear canal. Position them in the open air and use them to record sound. Now build a fake life-size human head using materials that approximate the density & reflectivity of human tissue and skin, and insert these same mics deep into the ear canals, facing outward. Now measure the same sound again. The difference between the two recordings is the HRTF of your dummy head.

Every person has an individual HRTF and the variance from person to person is significant. Since headphones bypass the HRTF, in order to sound natural they must have a frequency response that matches the HRTF. Put differently, a headphone with flat frequency response would sound quite dull, down 15+ db in the 2 – 5 kHz range.

This doesn’t apply to loudspeakers. If a speaker has objectively flat FR, every person will perceive that however they perceive natural sounds. Speakers don’t have to reproduce the HRTF because the sound comes from a distant source and your HRTF transforms it when it hits your body. Headphones play sounds directly into your ears, bypassing your body, head and HRTF.

This means there is an absolute reference FR for speakers: perfectly flat. But there is no absolute reference FR for headphones. A headphone has to mimic the HRTF which is different for every person. The best a well-engineered headphone can do is mimic the most common or average HRTF across the population. Each individual will be a little different.

Thus, different people will disagree on what headphone has the most natural FR reproducing sounds most realistically. For example, the Sennheiser HD-800 has a big response rise around 5 – 7 kHz. For me personally, it’s artificially bright, almost skull-jarring. But for others it may sound natural. At the opposite end of the spectrum, the Audeze LCD-2 has a dip from 2 – 9 kHz (its raw response has a rise, but it rises a bit less than the typical HRTF does). For me personally, it sounds natural and realistic. My HRTF probably lifts this frequency range less than average. But for others this headphone sounds dull.

Good IEMs (In-Ear Monitors) aka Earbuds

Most phones have very good audio output quality via their analog headphone jack <note to self: resist urge to crack iPhone 7 joke…> if you can find earbuds that sound good. Yet that’s a big if. Most earbuds sound like crap. Of the few that sound good, some have impedance or efficiency that don’t match well to a phone. Also, while phone audio output is often very good, that does not mean excellent or reference quality, so don’t go overboard and waste money on really expensive IEMs.

I listen to mostly natural acoustic music and I’m picky about sound. The best really good IEM I’ve found is the Vsonic GR07. They cost about $100 and sound really good. To my ears, they’re comparable to a pair of full-size HD-600. They have flat, neutral response that is neither warm nor bright, but just right. The treble is smooth with good detail. Due to a phone’s audio output limitations, even with uncompressed FLAC files the extreme high frequencies are rolled off and there’s less “air”, slightly less crisp transient response.  It sounds very good, even great, but not excellent or reference quality. The GR07 is about the best quality it’s worth paying for given the limitations of the source.

The best cheap IEM I’ve found is the Vsonic VSD1S. They cost about $35, have the same high quality construction and 90% as good sound as the GR07. Comparatively, the VSD1S has a slight midrange recess, not quite as smooth or extended treble. Overall, they still have a nice neutral sound despite have a touch more “V” shaped response curve. They’re good enough to listen to for hours enjoyably without fatigue, better than most other good IEMs, but just not quite as refined as the GR-07.

A Cheap Audiophile Headphone System

Here’s a cheap audiophile quality sound system:

That’s it. Connect the Juli@ unbalanced analog outputs to the amp’s inputs. Play your CDs, DVDs, whatever on the computer. Use whatever headphones you want.

Total cost: $510 = $170 for the card, $340 for the amp. Plus the headphones. You can get some vintage Sennheiser HD-580 or HD-600 on eBay for a couple hundred bucks. Or you can go all-in with a really nice set of headphones like the Audeze LCD. The Corda Jazz has a smooth sound, detailed and sweet yet neutral, with enough power to  drive almost any headphone on the planet.

I use this as a secondary system to drive my Audeze LCD-2F when my reference system is unavailable. It is amazing – 95% as good as the reference system. Extreme treble and large ensemble complex music is not quite as refined, but that’s just picking nits because it sounds damn good.

Years ago when I was in college I would have climbed a mountain of broken glass for sound like this, especially at this price.

Note: you can get an audio system like this even cheaper from JDS Labs. Get a single DAC+Amp for $300, so you don’t need the sound card. Just stream the bits from the USB port of any computer into the DAC. JDS Labs is the American version of Jan Meier’s Corda in Germany: a few guys in Illinois who are good electrical engineers and take a non-nonsense approach to building audiophile quality gear without audiophile prices or bullshit.

The Fantastic Audeze LCD-X

A few years ago I found the best headphones I’ve ever heard, the Audeze LCD-2. These are the 2014 Fazor version. A while later I made them even better with a subtle parametric EQ. That may sound like sacrilege to some audiophiles, but it works for me. The LCD-2 has  enhanced my late-night music listening and I still enjoy and use them regularly.

Since then, Audeze came out with another headphone: the LCD-X. It is designed to have a more neutral (flatter) frequency response and faster/cleaner transient response. Both of these claims are substantiated by measurements. But how do they sound? I wanted to find out. Audeze had a sale so I ordered a pair to get a listen.

Dimensionally, the X are exactly the same as my 2, or so close I couldn’t tell the difference. They’re black and made of metal, where the 2 are wood. The X are a bit heavier, but I didn’t feel the difference. Clamping force, fit, they felt exactly the same on my head.

I ran the comparison through my Behringer DEQ2496. More precisely, CDs played on my Oppo BDP-83, toslink to DEQ2496, toslink to Oppo HA-1, balanced headphone out. The DEQ2496 enabled me to level match within 0.5 dB, keeping the signal otherwise unchanged, or apply EQ as mentioned below. This doesn’t use the DEQ2496’s DA or AD converters; it operates in pure digital mode. Subjectively, I found the X to be 8.5 dB louder than the 2, so used this to equalize the levels.

Tech note: According to specs the X is about 11 dB louder than the 2 at the same volume setting. According to Audeze specs, the X makes 103 dB with 1 mW of power and has a 20 ohm impedance. Since it’s planar magnetic, the impedance is flat vs. frequency. That means 0.1414 V (141 mV) will make 103 dB, so 0.0317 V (32 mV) will make 90 dB. The voltage sensitivity of the LCD-2 is 0.114 V @ 1 kHz @ 90 dB. So we have 20*log(0.141/0.032) = 12.9 dB. My subjective impression was slightly different. Attenuated by 13 dB, the X was quieter than the 2; I used 8.5 dB.

First I did the fair comparison: head to head, no EQ. Here it was no contest: the X was easy to differentiate, and overall better sound:

  • X has more upper mid – no dip like the 2
  • X bass is slightly (about 2dB) quieter, but just as flat and deep
  • X has slightly better bass clarity
  • X has more linear and extended treble
  • X sounds “cool”, not “warm” like the LCD-2
  • Detail: X is on stage with the musicians, 2 is in the 5th row back
  • The X has more detail than reality; the 2 has less than reality; neither is perfect but the X is closer

However, I don’t listen to my 2s straight. I apply a parametric EQ: +3 dB @ 4600 Hz, Q=0.67 (3 dB / octave, 2 octaves wide). This counteracts the 2’s softness in the upper mids and lower treble, giving it a more neutral response curve and a bit more detail as if you’re sitting a few rows closer to the stage.

So next I did the realistic comparison: how I would actually listen to them: X raw, versus 2 with the above EQ:

  • They sound almost the same
  • X emphasizes the overtones, but still has the core sound
  • 2 favors the core sound, but still has the overtones
  • X is slightly more clear, yet less realistic voicing on some recordings
  • 2 has more realistic voicing on most recordings, yet slightly less clear
  • 2 is on the warm side of reality, X is on the cool side
  • Overall, which sounds better depends on the recording

Here it was a much harder decision. I also compared them to my speakers. They were about equally close to that sound, yet approaching it from opposite sides. These are both excellent headphones and I could be happy with either. They wipe the floor with any conventional dynamic headphone I have ever heard. If I didn’t already own the 2, or if I didn’t have a digital parametric EQ, I would pick the X. But I do already own the 2, and with the parametric EQ they are just as good as the X. I listen mostly to acoustic music and the 2’s realistic voicing is more important to me than the X’s extra 1% of detail. So why change anything?

I kept my LCD-2F and returned the LCD-X thanks to Audeze’s excellent service which includes a 30 day trial period. It was a fun experiment and satisfied my curiosity. While I kept my LCD-2F, I can heartily recommend the LCD-X to anyone who wants a fantastic set of headphones with dynamic and detailed yet realistic sound.

The Amazing Audeze LCD-2 (rev 2 Fazor)

A couple years ago I bought a pair of Audeze LCD-2 headphones. I’ve listened to many headphones over the years and they are the best headphones I’ve ever heard. This is what I had to say about them.

But, like all things created by mankind, they’re not perfect. Their near-perfect frequency response has a small dip between 2 kHz and 9 kHz. It’s linear and smooth, so subjectively is barely noticeable. Yet it slightly subdues the sound, as if you’re sitting a few rows back from the 1st row.

Since I recently got a digital signal processor, I figured I’d try it out on the headphones. I put a single parametric EQ, +3 dB, centered at 4,600 Hz, 2 octaves wide (slope 3 dB / octave, or Q=0.67), so it has effect between 2,300 and 9,200 Hz. To my ears, this made the LCD-2 absolutely perfect. It’s subtle yet definitely noticeable (I blind tested it on a variety of recordings), and shifts you back to the 1st row of the audience.

I tried +4 dB and it was good, though a bit more than needed. +2 was not quite enough; +3 was perfect. And I tried shifting the frequency up and down a bit, but 4,600 Hz sounded perfect.

From what I can see in specs, this makes the LCD-2 sound closer to the LCD-X, taking it from slightly warm or rounded, to neutral. The LCD-2 still sounds yummy, yet realistic – yet now it’s a touch more detailed. This EQ doesn’t change the character of the sound, it just makes that dip shallower giving a bit more upper midrange and treble detail. It’s about as close to perfect sound as human engineering can achieve in a headphone.

I’ve considered getting the LCD-X but this change nixed that entirely, making the LCD-2F near enough perfection to keep for a long time.

The Power of the Dark Side

First let’s cut to the chase: in-room far-field frequency response measured at the listening position using 1/3 octave warble tones, measured with a Rode NT1-A mic, corrected for mic response


  • The red line is what you hear – near perfection!
  • The solid blue line is with room treatments, but without EQ
  • The dotted blue line is without room treatment

In short, you can see that room treatment (huge tube traps and copious use of thick RPG acoustic foam) made a huge difference. Then EQ finessed that to something near perfection.

Now, for the details:

I’ve been an audiophile since my late teen years, long before my income could support the habit. As an engineer and amateur musician I always approached this hobby from a unique perspective. The musician knows what the absolute reference really sounds like – live musicians playing acoustic instruments in the room. The engineer believes objectivity – measurements, blind listening tests, etc. – is the best way to get as close as possible to that sound.

Part of this perspective is being a purist, and one aspect of being a purist is hating equalizers. In most cases, EQ falls into one of 2 categories:

  1. There are flaws in the sound caused by the speakers or room interactions, and instead of fixing them you use EQ as a band-aid. This flattens the response but leaves you with distortions in the phase or time domain, like ringing.
  2. You don’t want to hear what live acoustic music really sounds like, you prefer a euphonically distorted sound and use an EQ to get it.

Equalizers are the dark side of audio. Powerful and seductive, yet in the end they take you away from your goal: experiencing music as close as possible to the real thing. Recently I traveled to the dark side and found it’s not such a bad place. Share my journey, if you dare.

I had my audio room here in Seattle dialed in nicely after building big tube traps, thick acoustic foam and careful room arrangement based on repeated measurements. However, it still had two minor issues:

  1. A slight edge to the midrange. From personal experience I describe it as the sound I hear rehearsing on stage with the musicians, rather than being in the 2nd row of the audience.
  2. The deepest bass was a bit thin, with 30 Hz about -6 dB. I have a harp recording where Heidi Krutzen plays the longest strings, which have a fundamental around 25 Hz. I could hear this in my room, but it was a subtle whisper. It would be nice to hear that closer to a natural level.

My room treatments made a huge improvement in sound (and I have the measurements to prove it). But I don’t know of any room treatment that can fix either of these issues. The sound was very good both objectively (+/- 4 dB from 35 Hz to 20 kHz at listener position) and subjectively, and I enjoyed it for years. Then I got the LCD-2 headphones and Oppo HA-1 DAC. As I listened to my music collection over the next year (a couple thousand discs, takes a while), I discovered a subtle new dimension of natural realism in the music and wanted to experience that in the room.

Since my upstream system was entirely digital, equalization might not be as terrible as any right-thinking purist audiophile would fear. I could equalize entirely in the digital domain, no DA or AD conversion, before the signal reaches the DAC. And since the anomalies I wanted to correct were small, I could use parametric EQ with gradual slope, virtually eliminating any audible side effects.

That was the idea … now I had to come up with an action plan.

After a bit of Googling I found a candidate device: the Behringer DEQ2496. Price was the same on B&H, Adorama and Amazon, and all have a 30 day trial, so I bought one. The DEQ2496 does a lot of things and is complex to use and easy to accidentally “break”. For example, when I first ran the RTA function, it didn’t work. First, the pink noise it generates never played on my speakers. After I fixed that, the microphone I plugged in didn’t work. After I fixed that, the GEQ (graphic equalizer) settings it made were all maxed out (+ / – 15 dB). Finally I fixed that and it worked. All of these problems were caused by config settings in other menu areas. There are many config settings and they affect the various functions in ways that make sense once you understand it, but are not obvious.

NOTE: one easy way around this is before using any function for the first time, restore the system default settings, saved as the first preset. This won’t fix all of the config settings; you’ll still have to tweak them to get functions to work. But it will reduce the amount of settings you’ll have to chase down.

In RTA (room tune acoustic?) mode, the DEQ2496 is fully automatic. It generates a pink noise signal, listens to it on a microphone you set up in the room, analyzes the response and creates an EQ curve to make the measured response “flat”. You can then save this GEQ curve in memory. You have two options for flat: Truly flat measured in absolute terms, or the 1 dB / octave reduction from bass to treble that Toole & Olive recommend (-9 dB overall across the  band). This feature is really cool but has 2 key limitations:

  1. It has no built-in way to compensate for mic response. You can do this manually by entering the mic’s response curve as your custom target response curve, but that is tedious.
  2. It provides only 15 V phantom power to your mic. Most studio condenser mics (including my Rode NT1-A) want 48 V, but aren’t that sensitive to how much voltage they get and work OK with only 15 V. But you always wonder how much of the mic’s frequency response and sensitivity you lose when you give it only 15 V. Perhaps not much, but who knows?

The GEQ settings the DEQ2496 auto-generated were too sharp for my taste, so I looked at the FR curve it measured from the pink noise signal. This roughly matched the FR curve I created by recording 1/3 octave warble tones from Stereophile Test Disc #2. Since both gave similar measurements, I prefer doing it manually because I can correct for the mic’s response, and my digital recorder (Zoom H4) gives the mic full 48 V phantom power.

So the curves match: that’s a nice sanity check – now we’re rolling.

Using the DEQ 2496, I created parametric EQ settings to offset the peaks and dips. This enabled me to use gentle corrections – both in magnitude and in slope. I then replayed the Stereophile warble tones and re-measured the room’s FR curve. The first pass was 2 filters that got me 90% of the way there:

  • +4 dB @ 31 Hz, 1.5 octaves wide (slope 5.3 dB / octave)
  • -3 dB @ 1000 Hz, 2 octaves wide (slope 3 dB / octave)

These changes affected other areas of the sound, so I ran a couple more iterations to fine tune things. During this process I resisted the urge to hit perfection. Doing so would require many more filters, each steeper than I would like. It’s a simple engineering tradeoff: allowing small imperfections in the response curve allows fewer filters with gentler slope. Ultimately I ended up with near-perfect frequency response measured in-room at the listening position:

  • Absolute linearity: from 30 Hz to 20 kHz, within 4 dB of flat
  • Relative linearity: curve never steeper than 4 dB / octave
  • Psychoacoustic linearity: about -0.8 dB / octave downslope (+3.9 dB @ 100 Hz, -3 dB @ 20 kHz)

The in-room treble response was excellent to begin with, thanks to the Magnepan 3.6/R ribbon tweeters. Some of the first EQs impacted that slightly, reducing the response from 2k to 6k, so I put in a mild corrective boost.

Subjectively, the overall before-after differences are (most evident first):

  • Midrange edge eliminated; mids are completely smooth and natural, yet all the detail is still there.
  • Transition from midrange to treble is now seamless, where before there was a subtle  change in voicing.
  • Smoother, more natural bass: ultra-low bass around 30 Hz is part of the music rather than a hint
  • Transition from bass to lower midrange is smoother and more natural.

In other words, audiophile heaven. This is the sound I’ve dreamed of having for decades, since I was a pimpled teenager with sharper ears but less money and experience than I have now. It’s been a long road taken one step at a time over decades to get here and it’s still not perfect. Yet this is another step toward the ideal and now about as close as human engineering can devise. The sound is now so smooth and natural, the stereo stops reminding me it’s there and enables me to get closer to the music, which now has greater emotional impact. And it’s more forgiving of imperfect recordings so I can get more out some old classics, like Jacqueline DuPre playing Beethoven Trios with Benjamin Britten and Arthur Rubinstein playing the Brahms F minor quintet with the Guarneri.

Throughout this process, I could detect no veil or distortion from the DEQ2496. The music comes through completely transparently. I measured test tones through the DEQ2496 in both pass-through and with EQ enabled; it introduced no harmonic or intermodulation distortion at all. That is, anything it might have introduced was below -100 dB and didn’t appear on my test. This is as expected, given that I’m using it entirely in the digital domain – no DA or AD conversions – and my EQ filters are parametric, small with shallow slope.

While I was at this, I created a small tweak for my LCD-2 headphones. Their otherwise near perfect response has a small dip from 2 to 8 kHz. A little +3 dB centered at 4.5 kHz, 2 octaves wide (3 dB / octave, Q=0.67) made them as close to perfect as possible.

Overall, I can recommend the DEQ2496. Most importantly, it enabled me to get as close to humanly possible to perfect sound. That in itself deserves a glowing recommendation. But it’s not a magic box. I put a lot of old fashioned work into getting my audio system in great shape and used the DEQ2496 only to span that last %. Like any powerful tool, the DEQ2496 can be used for evil or for good. So to be fair and complete I’ll list my reservations:

  • The DEQ2496 is not a magic band-aid. You still need to acoustically treat and arrange your room first to fix the biggest problems. After you do that, you might be satisfied and not need the DEQ2496.
  • The DEQ2496 is complex to use, creating the risk that you won’t get it to work right or you’ll get poor results.
  • To use the RTA feature you’ll need an XLR mic with wide, flat frequency response.
  • I cannot assess its long term durability, having it in my system for only a few days. Many of the reviews say it dies after a year or two,  but they also say it runs hot. Mine does not run hot, so maybe Behringer changed something? Or perhaps mine runs cooler because I’m not using the D-A or A-D converters. It does have a 3 year manufacturer warranty, longer than most electronics.

Ubuntu Linux and Blu-Ray

Getting Linux to work with Blu-Ray took some custom configuration. The state of Linux and Blu-Ray has much to be desired and doesn’t work out of the box. But it can be made to work if you know what to do. Here’s how I got it to work.

Reading Blu-Rays

This was the easy part. You can do it in 2 ways: VLC and MakeMKV

Blu-Rays don’t play in VLC because of DRM. To play them in VLC you need to download a file of Blu-Ray keys, like here: This may not be the best approach because the file is static. New Blu-Rays are coming out all the time. But it works if you regularly update this file and it has the key for the Blu-Ray you want to play.

MakeMKV is software that reads the data from a Blu-Ray and can write it to your hard drive as an MKV file. It can also stream the Blu-Ray to a port on your local machine. Then you can connect VLC to play the stream from that port. Viola! You can watch the Blu-Ray on your computer with VLC, even if you don’t have the keys file. MakeMKV is shareware – free for the first 30 days, then you should pay for it.

Writing Blu-Rays

The first challenge writing Blu-Rays is Ubuntu’s built-in CD writing software, cdrecord. It’s a very old buggy version. This happens even with the latest repos on Ubuntu 15.10. It works fine for Audio CDs, data CDs and DVDs. But not for Blu-Ray. The first step is to replace it with a newer, up-to-date version. The one I used is CDRTools from Brandon Snider:

Whatever front end you use to burn disks (like K3B) works just the same as before, since it uses the apps from the underlying OS, which you’ve now replaced. After this change I could reliably burn dual-layer (50 GB) Blu-Rays on my Dell / Ubuntu 15.10 desktop using K3B. My burner is an LG WH16NS40. It is the bare OEM version and works flawlessly out of the box.

Now you can burn a Blu-Ray, but before you do that you need to format the video & audio and organize into files & directories that a Blu-Ray player will recognize as a Blu-Ray disc. What I’m about to describe works with my audio system Blu-Ray player, an Oppo BDP-83.

The command-line app tsmuxer does this. But it’s a general transcoder that can do more than Blu-Ray, and the command line args to do Blu-Rays are complex. So I recommend also installing a GUI wrapper for it like tsmuxergui.

sudo apt-get install tsmuxer tsmuxergui

Now follow a simple guide to run this app to create the file format & directory structure you need for a Blu-Ray. Here’s the guide I used. Do not select ISO for file output. When I did that, K3B didn’t know what to do with the ISO – my first burn was successful, but all it did was store the ISO file on the disk. Instead select Blu-ray folder. This will create the files & folders that will become the Blu-Ray. Also, you might want to set chapters on the tsmuxer Blu-ray tab. For one big file that doesn’t have chapters, I just set every 10 mins and it works.

When tsmuxer is done, run K3B to burn the files & folders to the blank Blu-Ray. Key settings:

In K3B:
Project type: data
The root directory should contain the folders BDMV and CERTIFICATE
Select cdrecord as the writing app
Select Very large files (UDF) as the file system
Select Discard all symlinks
Select No multisession

Then let ‘er rip. Mine burns at about 7-8x, roughly 35 MB / sec. When it’s done, pop the Blu-Ray into your player and grab some popcorn!

Review: Oppo HA-1


I’m quite happy with my audio system. Why change anything?

My audio system is analog based. That is, each source component has a line level analog output that drives my poweramp/speakers, and my headphone amp. I built a 10k passive attenuator to select the source and set volume levels. Having a single metal film resistor in the signal path, it gives the ultimate sonic transparency. No active preamp of any design at any price can be more transparent.

However, that doesn’t mean better sound cannot be had. Lately I realized my phono amp is my only true analog source. All other sources – from tuner, to Tivo, to CD player, etc. were all digital and offered digital audio outputs (optical or coax). Each of these devices also has an analog output, which means it has its own on-board DAC and analog output stage.

What if I could take the raw digital output of each source, bypassing its on-board DAC and analog output stage, and send this data stream to a dedicated DAC? Having a single central DAC enables one to invest in reference quality, bypassing the cheaper lower quality DACs and analog stages built into each source device. The potential benefit would be having a single reference quality DAC and analog output stage used for all my source devices. Another benefit would be the source devices become less critical for sound quality. I won’t need a dedicated CD player anymore; the DVD player plays CDs and its digital output is indistinguishable from the CD. It’s the on-board DAC and analog output stage that differentiates the best sounding players, and I won’t need that anymore.


So what kind of DAC would I consider?
First some simple requirements:

  • Reference Quality: DAC, analog line stage, and headphone amp
  • Digital Inputs: optical, coax and USB. I already have a 6-way switch where each of the 6 inputs can be RCA unbalanced, Toslink, and Component Video
  • Analog Input: need this for my phono amp (I occasionally listen to and record LPs to CD)
  • Analog Output: need 2 of these – one for my power amp, one for my Tascam recorder
  • Headphone amp: to drive LCD2-F and HD-580s

Most DACs meet these requirements. The only 2 sticklers are (1) Reference Quality and (2) Analog input.

The candidates came down to 2 units:

  • Grace m903 (or m920)
  • Oppo HA-1

Other good units like the Benchmark DAC1 (or DAC2) were out due to their lack of analog input, or lack of line level analog outputs.
Ultimately it came down to the Oppo HA-1 because it is the only unit that has all of the following:

  • Single ended class A analog stage
  • Sabre ESS9018 DAC
  • Pure balanced operation: both headphone amp and analog line out (though it also has unbalanced outputs for both
  • Cost (half the price of the Grace)

If the HA-1 didn’t pass muster, I’d give the Grace a try. One Grace feature I’d like to have, that I gave up with the Oppo, is a headphone crossfeed circuit. This sounds gimmicky, but the Grace uses Jan Meier’s crossfeed, which I have come to enjoy in the Corda Jazz headphone amp I use with my desktop computer audio system.

The Sound

The HA-1 sounds a lot like the Corda Jazz, which I reviewed here. This is a good thing. It’s clean, neutral, and natural sounding. The Maxed out Home, in comparison, is a bit warmer and the extreme highs are a bit softer. The Maxed out Home is an excellent amp, but I prefer the Jazz – and the HA-1.

The HA-1 headphone output level is 6 dB louder (twice the voltage, 4x the power) in balanced mode. The HA-1 is designed fully balanced from the digital to analog stages, and provides the cleanest, lowest distortion sound in that mode.

The HA-1 headphone output has two gain modes: low and high. This gives it flexibility to drive any headphon with any kind of music. I’m using the HA-1’s low gain, mainly because I’m using the balanced output which is 6 dB louder than unbalanced. In unbalanced mode, into my LCD2-F and HD-600 headphones, low gain mode isn’t quite enough for recordings made at low average levels (like music with wide dynamic range, recorded quiet to have room for dynamic peaks).

Speaking of the line stage. I’m using the HA-1 balanced line outs to drive the balanced inputs of my Adcom 5800. The Magnepans in my listening room are just a touch more transparent and revealing than the LCD-2 Fazor, though the Mags roll off below 30 Hz and LCD-2 doesn’t. The comparison here is the analog output of my CD player, through the passive attenuator driving the unbalanced inputs of the 5800 (what I’ve been listening to for years), versus the digital output of the CD player, sent to the HA-1, driving the balanced inputs of the 5800. So to switch I had to turn off the 5800, flick switches on the back from unbalanced to balanced, then turn it back on. This takes about 30 seconds, too long to be ideal for comparison, but it’s the best I could do.

Unsurprisingly, they sound quite similar. In order from most evident to least: the HA-1 is a touch faster/cleaner in the extreme high frequencies, a touch lighter in the bass, and a touch smoother in the midrange.

High Frequencies

In the highs, it took a high quality recording of castanets, and harp, to hear that the transient response was just a smidge faster and cleaner on the HA-1. My prior system was no slouch, but there’s no question the HA-1 does extreme high frequencies better. I’m talking about frequencies high enough we don’t hear them as “treble”, but we hear them as “air” or cleaner transient attack. The strange thing is, the HA-1 is not only a touch faster and cleaner, but also lighter. The castanets and sharp harp plucks on the short top register strings weren’t louder, just better defined, portrayed with a lighter more natural sound.

Low Frequencies

Listening to acoustic percussion (Rabih Abou-Khalil’s Tarab, Fredericksen’s Elfin Knight), the HA-1 portrays the bass with a lighter touch – overall there’s simply just a bit “less bass”. But the HA-1 bass goes just as deep. This evident on harp recordings – big harp strings can emit incredibly deep bass with frequencies below 30 Hz. It’s subtle but moves the air in the room. The HA-1 loses nothing – all those ultra-low resonances are there. So while the HA-1 overall presentation is slightly “leaner”, the ultra-deep bass is all there. Then for fun I popped in Diana Krall “Temptation”. Fantastic – the bass was phenomenal, her voice was amazing, and the percussion transient attacks were fast and light.


With the mids, with only a couple of hours of listening I couldn’t decide whether the smoothness of the HA-1 is a slight veil obstructing detail, or whether that extra detail was artificial (perhaps a very slight amount of intermodulation distortion). It may be a bit of both. For example when Cecelia Bartoli belts out the crescendos in Rossini’s Belta Crudele, the smooth core of her voice has an spine chilling edge to it, which I used to think was just getting too close to the mic. With the HA-1, that edge is still there but smoothed just a touch. More natural, I think so. But there was some detail that is missing – whether that detail was in the original recording, or distortion, was an open question resolved by further listening.

Voicing: the HA-1 mids are more open and natural; the CD player’s analog output in comparison is ever so slightly congested. Put differently, the CD player has a midrange “presence” that the HA-1 does not. If you heard the CD player alone you’d never say it has congested mids – it sounds great. Only in direct A/B comparison does this subtle difference become evident. This is most noticeable on some recordings that have a bit of congested midrange, for example the Chieftans 7, or to a lesser extent, Wincenc & Raps Mozart flute/piano sonatas on Naxos. The congested midrange is a flaw in these recordings. The CD player’s analog outputs accentuate this flaw, while the HA-1 opens up the congestion a bit, revealing a bit more detail and sounding more natural. Interestingly, one of the reasons I picked this CD player (Onkyo DX-7555) is because its mids were more open and natural, less congested than the Rotel RCD-1070 I had been using before. Once again, the HA-1 is a similar improvement taken to a higher level.

Resolution: Large Ensemble Music

Large ensemble music (10 or more acoustic instruments playing simultaneously) really differentiates the HA-1 (from the CD player analog output through the attenuator). The HA-1 simply resolves it better. Each instrument is more individually identifiable, sounding less like a “wall of sound”. What’s really interesting is, 15 years ago when I built the 10k attenuator, this is one area that really differentiated it from the preamp I had been using. Back then, I figured this was the result of the elimination of intermodulation distortion. The HA-1 is a similar improvement, taking it to a new level (the difference isn’t huge, but it is clearly audible). To be clear, the HA-1 is not “brighter”, it’s just more clear, natural and airy sounding with large ensemble music. This is most likely the result of more accurate response from 10 kHz upward.


These differences are subtle, but easily audible. Put together, they’re enough to make the HA-1 a worthwhile addition to my system. It’s simply excellent all around – the DAC, the headphone amp, and the line stage. It’s superbly transparent and neutral – not warm, not cold, not harsh, nor lush. It sounds like whatever recording you’re listening to. It’s ultra detailed yet the detail is natural without a hint of harshness or midrange edge. Big dynamics are effortless and microdynamics add to the realism.

Some have given the HA-1’s linestage mixed reviews. I have no mixed feelings about it. It’s fantastic, better than any active preamp I’ve ever heard, with transparency rivaling my 10k ladder stepped attenuator. Mag 3.6/R speakers are revealing and unforgiving, and the HA-1 really makes them sing.