Having owned Magnepan 3.6/R for 20 years and set them up in 3 very different listening rooms, I’ve learned a few things. I want to capture the important things here.

Overview

Definitions:

• Front wall: in front of the listener, behind the speakers.
• Rear wall: behind the listener, in front of the speakers.
• SBIR: speaker boundary interference response
  • The total response at the listener position includes sound reflected from the front and side walls near the speaker.
  • This response depends on the distance and angle of the speaker to these walls, and the treatment of those walls.
• LBIR: listener boundary interference response
  • The total response at the listener position includes sound reflected from the rear and side walls near the listener.
  • This response depends on the distance and angle of the listener to these walls, and the treatment of those walls.
• Speed of sound: 1130 f/s at sea level and 70*. Slower when cold, faster when warm.

All speakers are sensitive to room setup, but planars are dipoles which are more sensitive than conventional speakers. This is both a blessing and a curse. The blessing: if something isn’t right you can often fix it with simple rearrangement. The curse: for ideal sound, the speakers are going to be further into the room away from the walls.

SBIR

All speakers (even forward-firing cones) propagate both forward and back. But a dipole’s back wave has inverted amplitude.

Note: inverted amplitude is is often called or 180* out of phase, which is misleading. 180* out of phase means a shift, while inverted means a flip. Music has many frequencies superimposed so one wonders, 180* out of phase at what frequency? It is more precise to call it an amplitude inversion. More on this here.

Example 1: consider a speaker parallel to the front wall, 3′ away, which is 1/4 wavelength of 94 Hz. The back wave hits the front wall, reflects and as it passes the speaker it has traveled 1/2 wavelength, so it is 180* out of phase with the direct (non-reflected) wave from the speaker. This attenuates 94 Hz. But if the speaker is a dipole, it does the opposite (boosts) because the back wave started out with inverted amplitude, so shifting it 180* out of phase brings it back in-phase.

Conclusion: due to SBIR, dipoles boost the 1/4 wavelength frequency.

Example 2: consider what that same speaker does at 188 Hz (twice the frequency, half the wavelength). Now the 3′ distance is 1/2 wavelength, so the distance traveled is a full wavelength. A conventional speaker will boost this frequency because it’s in phase. A dipole will cut this frequency.

Conclusion: due to SBIR, dipoles cut the 1/2 wavelength frequency.

Direct vs. Reflected

Dipoles (electrostatic or planar magnetic) have a flatter impedance vs. frequency curve, without the strong Q resonances that conventional speakers have. This makes them a near-resistive load which is easy for amps to drive and gives them flatter phase response and group delay. I believe this contributes to their big, open, transparent sound relative to conventional speakers which can sound thick and muddy in comparison.

With all speakers, the sound you hear is a mix of direct and reflected. With dipoles this mix has relatively more reflected, less direct. This can make them sound big and phasey in underdamped rooms. With dipoles your room typically needs more damping than it does with conventional speakers.

One way to tackle this is to damp the walls to reduce reflection. How much damping you need and where to put it depends on the room size, shape, materials, and your personal preference. Too much damping and the dipole will sound thick & muddy like a conventional speaker.

Some dipoles (like Magnepans) have a rise in bass response that is supposed to be attenuated by the back wave reflected from the front wall. Because of this, they need to be the right distance from the front wall, and you don’t want to damp that wall too much.

Conclusion: in small to medium sized rooms, you will need to damp the wall behind dipoles to some extent, but not entirely. This damping must be effective down into bass frequencies, so it can’t just be acoustic foam; it must be tube traps, bass traps, etc.

LBIR

This topic doesn’t at first appear to be unique to dipoles, but it turns out to have an important difference. Consider a listener 3′ in front of the rear wall. Sound from the speakers reflects from the rear wall and comes forward, having traveled 6′ when it reaches the listener again. At 94 Hz, this is half a wavelength, so it attenuates that frequency. At 188 Hz this is a full wavelength, so it boosts that frequency.

What’s different about dipoles: the LBIR and SBIR distances, when equal, negate each other’s effects. With conventional speakers, they exaggerate each other. That is: if the speakers are 3′ from the front wall and the listener is 3′ from the back wall, the reflected waves don’t affect frequency response; SBIR cuts the same frequencies that LBIR boost. Conventional speakers give a double-sized cuts and boosts at the same frequencies.

Conclusion: when setting up dipoles in a small to medium sized rooms, try to make the LBIR and SBIR distances roughly equal. Put differently: the distance from the listener to the back wall should be the same as the distance from the speakers to the front wall.

More specifically, why I like planar magnetic speakers (and headphones!).

Sound quality: this one is subjective, yet important. When set up properly, planars sound more natural, open, and transparent than conventional speakers. They’re perfect for acoustic music across all genres from small to large ensemble classical, jazz, vocals, etc. Solo piano, vocals and chamber music are particularly good on planars. Their midrange is uncolored, having incredibly high resolution, yet without the artificial detail of boosted upper mids/treble, and without adding the glare or edginess of conventional dynamic drivers — unless that edginess is in the recording itself! With the 3.6/R I frequently hear subtle musical details or tone/balance shifts that I never hear even on the best headphones. Music is mostly midrange, and that is what planars do best. And the treble is simply astounding. No speaker on Earth matches the high frequency extension and linearity of those huge ribbon tweeters. The transition from the mid panel to treble ribbon is seamless, preserving the timbre and harmonic structure of acoustic instruments and voices. And that bass… clean, tight, with a seamless linear transition from the mids.

Low distortion: Measuring total distortion in Room EQ Wizard, my  Magnepan 3.6/R measure about -60 dB (0.1%) in the treble, -50 dB (0.3%) in the midrange, and -40 dB (1%) in the bass (at 60 Hz). That’s lower than most conventional speakers, even lower than most headphones. And it is an uncorrected figure, including the distortion in the microphones, amplifier, and DAC; the actual distortion from the speakers alone is even lower. The Audeze LCD-2 headphones (planar magnetic) measure < 1% total distortion throughout the entire frequency spectrum, even to sub-bass frequencies. No conventional headphone matches that, not even the Sennheiser HD-800.

Why is planar distortion so low? I can think of 2 reasons. First, each Mag 3.6 panel spans the area of about six 12″ woofers, and its ribbon tweeter is 5′ long. Such physically large drivers take only very small movement/excursion to produce a given sound level. And the distortion that a driver produces is related to its excursion. Second, the drivers don’t have as strong Q resonances as conventional drivers do, both mechanical and electrical.

Linear phase: The 3.6/R have a relatively flat impedance curve: 4.2 ohms in the bass, to 3.3 ohms in the treble. They don’t have the big impedance vs. frequency swings that conventional speakers have. This promotes linear phase and flat group delay.  The 3.6/R measure group delay of a flat zero through most of the frequency range, and only exceeds 10ms in the bass (below 80 Hz).

Easy load: Because planars have relatively flat impedance vs. frequency, they are primarily resistive loads that are easy for amplifiers to drive, despite their lowish impedance.

Drawbacks

Planars are dipoles, so they radiate equal energy front and rear, and the rear energy has inverted phase. This makes them more sensitive to room setup than conventional speakers. This can be a blessing or a curse, depending on your situation.

Planars tend to be inefficient, so they require more power for the same listening level. However, their dispersion is line-source (rather than a point-source), so the volume does not drop with distance as quickly as with conventional speakers.

Planars have limited maximum loudness. In a medium-large listening room, the bass distortion of my 3.6/R begins to rise at 95-100 dB SPL (and requires 400+ watts per speaker to attain). This is plenty loud enough for me, but it’s not for those who listen at ear-shattering levels.

Planars are difficult to measure because near-field, you can’t “hear” all the drivers from a single microphone position. And far-field, what you measure is as much the room as it is the speakers.

Planar drivers are side by side (the panel and the ribbon tweeter). They can’t be aligned vertically like conventional speakers, so the midrange to treble timing and impulse response depends on the angle between the speakers & listener. More specifically, the speakers should be angled so the panels are about 2″ closer to the listener than the ribbon tweeters.

Planars usually require a big room, and sound best when placed well into the room away from the walls. This leads to a low wife-approval-factor, and requires a dedicated audio room.

While planars have taut, low distortion bass, they usually don’t reproduce the lowest octave. The larger ones, like the 3.6/R, are good down to about 30 Hz, and 25 Hz is clearly audible though attenuated, which is fine for most music. But if you want that room-shaking 20 Hz rumble for movies with explosions and such, you’ll need a subwoofer.

Meier Audio has a feature in their amps called “FF” or Frequency Adaptive Feedback. Jan Meier describes it here. His article is detailed yet long. I wrote this article to complement it to help in understanding.

Musical Hearing

When it comes to human perception of sound and music, all frequencies are not created equal. The ear is most sensitive to frequencies from around 600 to 3000 Hz. And, most music (at least voices and acoustic music) is concentrated in this range.

Consequently, this is the most critical range for reducing distortion. You probably cannot hear 1% (-40 dB) distortion at 60 Hz, but you can hear it at 2000 Hz.

Analogy: Dolby B and RIAA equalization

Readers with a few grey hairs remember cassette tapes and Dolby B noise reduction from the 1970s and 80s. Dolby B was brilliant in its simplicity. Tape hiss has a wide frequency spectrum but it’s most noticeable in the treble (this is where our hearing is most sensitive). If you cut the treble during playback, it reduces hiss but it also dulls the music. So when recording, boost the treble. Then during playback, cut the treble by the same amount you boosted it. You get the same hiss reduction without any reduction in treble, because you’re only cutting exactly what you boosted earlier. The music has flat frequency response and sounds cleaner with higher S/N ratio.

The RIAA curve does the same thing for LPs. The pre-emphasis equalization curve cuts the bass relative to the treble before cutting the record groove. This reduces the groove and needle excursion needed to handle low frequencies, reducing distortion and noise. On playback, the phono head amp applies the opposite de-emphasis equalization curve, restoring flat frequency response.

The main drawback to this is that boosting the treble when recording limits the dynamic range. You can only boost it so far, before it reaches peak levels and overloads. Boosting the treble may require you to reduce the overall recording level. Alternately, reducing the bass lowers the SNR of the bass. Yet it improves the SNR of the treble, and this is a desirable tradeoff since that is where our hearing is much more sensitive to it.

Amplifier Feedback

Solid state amplifiers have a negative feedback loop that reduces distortion, increases bandwidth, and increases stability. Contrary to what we may read in some audiophile circles, negative feedback is A GOOD THING.

What exactly is negative feedback? An opamp’s native or open loop response, gain-bandwidth curve or transfer function, is not linear in both frequency and amplitude. So a portion of its output signal is inverted and fed back into the input, which offsets these non-linearities.

Furthermore, an opamp’s open loop response drops with frequency, around 20 dB per decade or 6 dB per octave. This means negative feedback has much stronger low frequencies than high frequencies. We can quantify this. Human hearing from roughly 20 Hz to 20 kHz spans a frequency range of 1000:1, or about 3 decades. So negative feedback is roughly 60 dB stronger at 20 Hz, than at 20 kHz.

More on negative feedback here.

This means most of the benefits of negative feedback are focused in the low frequencies. Higher frequencies have progressively less negative feedback. But perceptually, we want the opposite! Distortion & noise are much easier to hear in the high frequencies. So applying a pre-emphasis curve to the signal, similar to what RIAA does for vinyl, can be beneficial in the gain-feedback loop.

Frequency, Energy and Amplitude

Most of the amplitude in a musical signal is in the low frequencies. The midrange and treble, where our hearing is most sensitive, is just a smaller ripple riding on the much bigger bass wave. Reducing the amount of bass shrinks the entire signal, without any loss of amplitude or resolution in the midrange and treble. This keeps the signal away from the near-full-scale amplitude swings where devices get less linear.

This is particularly true of DACs – they get less linear for near-full-scale signals. Reducing the amount of bass before D to A conversion, then boosting it back afterward, can reduce distortion by keeping the DAC operating in its most linear region.

Frequency Adaptive Feedback

Combine these 4 ideas and you have Meier Audio’s FF. Start with the musical signal.

• Step 1: pre-emphasis: boost the critical frequency range (midrange-treble)
  • Alternately, attenuate frequencies outside this range. This can be a better approach since attenuation means no chance of clipping.
  • This is the first thing you do when the signal enters the amp.
• Step 2: pass this emphasized signal through the amp’s gain-feedback loop
  • Or through the DAC for D to A conversion
  • This weights negative feedback effects toward the critical frequency range
  • This reduces the signal from near full scale to the DAC’s more linear region
• Step 3: de-emphasis: attenuate the critical frequency range
  • Do the reverse of what you did in step 1.
  • This is the last thing you do before the signal leaves the amp.

In summary, FF has 2 potential benefits:

• Compensate for negative feedback’s bass-heavy content, giving relatively more correction at midrange/treble frequencies
• Reduce signal level to stay below peak levels having higher distortion, without reducing midrange/treble resolution

FF can be particularly effective for modern recordings which use heavy dynamic range compression with peak levels near full scale, or even have intersample overs or clipping.

Incidentally, the Redbook CD specification has something called “emphasis”, which is similar to FF. It boosts high frequencies (from 1 khz to 20 kHz). CD players are expected to attenuate those frequencies on playback. This is akin to Dolby B for digital audio.

Counterarguments

Here we’ll play some devil’s advocate.

If distortion is already below audibility, then FF is a solution looking for a problem – what is the point? In fact, the cure could be worse than the disease! FF requires filters on the input and output to shape the frequency response. These filters cause their own distortions (such as phase shift from analog filters or minimum phase digital filters). The overall effect is a trade-off between the benefits of FF and the drawbacks of having this extra signal processing.

Most opamps have far more gain than we need, so we must use a lot of negative feedback. So much, that the bandwidth is several times wider than audio, 100 kHz or more. Thus, even high frequencies have enough negative feedback to reduce distortion below audible levels, even if they have less feedback than low frequencies.

FF actually increases distortion outside the critical frequency range! With FF you will have higher distortion at lower frequencies (because FF attenuates them in the feedback loop). But you’ll have lower distortion in the midrange and treble. FF shapes distortion to match the sensitivity of our hearing: less distortion where our hearing is most sensitive, at the cost of higher distortion at low frequencies where we can’t hear it.

Today I learned how to use Room EQ Wizard to tune my audio room. I had already done room tuning on my own and was happy with the results. But REW enabled me to get it even better.

Here’s the final FR measured from the listener position, psychoacoustically smoothed. The Grey line is without EQ, the red is with EQ. You can see that the EQ is only a few bands from 500 Hz and lower.

It’s linear and smooth, with a typical tapering response. This is a treated room having big tube traps, bass traps, bass resonators and acoustic foam. The parametric EQ is mild with gentle amplitudes and slopes. I’d rather have a few little bumps in the response, than perfectly flat response with bloated phasey sound from extreme EQ settings. Don’t let the cure be worse than the disease!

Overall, this smoothed response throughout the range. During test listening I can switch curves instantly while the music is playing. My ears like the difference, especially noticeable on good acoustic music recordings.

Equipment & Details

• Test audio files created by REW version 5.2 beta 4, burned to DVD-A
• Oppo BDP-83 toslink PCM output
• Behringer DEQ2496 digital EQ, toslink input and output
• Corda Soul DAC-preamp, toslink input, XLR output
• Adcom 5800 amp (28 years old), XLR input
• Magnepan 3.6/R speakers (20 years old)
• Room treatments (floor-ceiling tube traps, RPG acoustic foam, etc.)
• MiniDSP UMK-1 measurement mic, and Rode NT1A mics, both calibrated
• Recorded from the listener position

Here are the rest of the REW plots:

Total distortion averaged about -50 dB (0.3%); higher in the bass, lower in the treble. That seems surprisingly low, considering it’s measured at the listener position and includes all distortion from the power amp, microphone & recorder. Many headphones, even some tube amps, have more distortion than this. These speakers reveal that the NT1A mics have lower distortion than the UMIK-1.

The bad news is that distortion at 40 Hz is about 10%. Yikes! But it’s down to 1% by 50-60 Hz, which would be great for headphones, quite rare for speakers.

I’ve always been happy with the bass response in this room after I treated it. 25 Hz is audible, if attenuated. But seeing these measurements, it seems that getting a subwoofer to handle everything below 60 Hz could “unload” the Magnepans and reduce overall distortion. I don’t want more bass, but tighter cleaner bass is always A GOOD THING.

Group Delay is pretty flat. Rises in the bass as usual. But it’s 10 ms or less from 60 Hz on up, a near perfectly flat zero for most of the range. This seems typical of planar speakers.

Initial impulse response is near zero in about 3 milliseconds, and you can see the reflections at 5 and 10 ms.

Total impulse energy is about -40 dB in the first 100 ms, from the listener position which includes room reverb. Room treatment damps the room, but it’s not completely dead. The grey is minimum phase IR, which is very close to the actual response.

The CSD looks linear (no obvious ringing frequencies above the bass region) and decently fast. The room treatment certainly helps here:

Here’s the Spectrogram. There’s some mild ring around 64 Hz and rising decay time below 50 Hz. Overall pretty flat and even. That’s room treatments doing their job.

Since I treated this room I’ve been happy with the sound. With these measurements I was able to apply EQ to fine tune some things I couldn’t fix with room treatments.

Generally speaking, balanced and differential signaling are two different things. They’re often (but not always) used together, and in audio, the term “balanced” refers to this.

Speakers and Headphones

A speaker or headphone responds to the voltage difference between its 2 input wires. It doesn’t assume either is ground, though one might be, it doesn’t matter. So connecting a speaker or headphone to a balanced output is easy. Just wire (-) to (-) and (+) to (+) whether or not the (-) is a ground (unbalanced output) or carries a signal (balanced output). If the unbalanced output has a common ground for both channels (like a headphone), you can split it to both L and R (-) in parallel.

Converting a balanced speaker or headphone output to an unbalanced connector is not as simple. An unbalanced headphone cable (a standard 1/4″ or 1/8″) has 3 wires: L (+), R (+), and a single wire that is a common ground for the L and R. You can’t connect a balanced output’s (-) wires to this ground. That would mix the channels, and allow the amp’s output stages to drive each other, which is bad because they usually have very low output impedance, so it can overdrive the output stages. Also, you can’t just ignore the output’s (-) wires and connect the headphone (-) wires together; this will give a common floating ground. In short, you need a transformer to do this conversion.

Components

If the balanced/unbalanced conversion is between components like a preamp (not a speaker or headphone), it gets more complex because unbalanced components assume the (-) is a ground, but the balanced (-) carries a signal and its ground is a separate (3rd) wire. You can’t connect a balanced output (-) signal to ground; it will overdrive the balanced output as it tries to swing a voltage over a 0 ohm load. Also, you need to ensure the (-) wire has the same impedance to ground as the (+) wire.

So the best way to convert unbalanced to balanced between components is to use a transformer.

However, you can wire unbalanced output directly to balanced input. Connect the unbalanced (-) output to both pins 1 and 3 on the balanced side (negative & ground), and the unbalanced (+) output to pin 2 on the balanced side (positive). That is, carry the unbalanced source ground through to the balanced input. Since unbalanced (consumer) output is at a lower voltage than balanced (pro), the downstream balanced component will be receiving a lower level signal than it expects. This may or may not be a problem, depending on how clean is the input signal and the balanced device’s input voltage sensitivity and gain.

This is part 8 of an 8 part series comparing the Meier Corda Soul and Oppo HA-1. Click here for the introduction.

Conclusion

Subjective Listening Impressions: Soul

• They sound similar which is expected for DACs/preamps that are well engineered with excellent specs.  Both are very neutral, transparent DACs. If you’re looking for euphonics, look elsewhere!
• However, the degree of similarity surprised me. I had to listen extremely carefully to specific recordings that I know well, to hear reliable differences. And even then, the differences were subtle.
• The differences were easier for me to hear on speakers. I suspect this is because my speakers are more neutral and resolving than my headphones.
• Speakers more resolving than headphones are rare, so most people, especially those with revealing headphones that are harder to drive (like the HD-800), will hear differences more easily on headphones than on speakers.
• To characterize the differences is to overstate them. But here they are:
  • Oppo: Earthy, Organic, Airy
  • Soul: Pure, Taught, Resolving
• Detailed summary of audible differences:
  • HF: Oppo has a touch more air; Soul has equal extension but less air. The first impression is slightly less HF from the Soul, but on deeper listen it is all there, yet less subjectively emphasized.
    • Ultimately, “all there but less emphasized” seems truer to live acoustic music, though different from what we normally perceive as “HiFi”.
    • Is “air” a barely perceptible hiss or noise that accentuates detail through stochastic resonance? If so, it’s a double-edged sword.
    • NOTE: “air” in the recording itself, like hearing the space in a good cathedral recording, is all there with both Soul & Oppo.
  • Treble: the Soul treble is smoother, making the Oppo sound slightly grainy in comparison. Though I would not say Oppo has grainy treble. The Soul’s treble response is unique in its naturalness.
    • Also: they balance the fundamental against harmonics slightly differently; Oppo emphasizes harmonics, Soul emphasizes fundamental. Each is a only a subtle variation of difference, both have uncolored voicing, and which sounds most natural depends on the recording.
  • Mids: Oppo is earthy or slightly “dirty”, with a hint more presence that adds a sense of extra detail in some recordings, slightly veiling in others. Soul sounds more transparent and pure, normally a good thing, though with some recordings sounding “sterile”.
    • The Soul has slightly greater midrange clarity.  It never revealed a musical detail the Oppo completely obscured, but it occasionally surprised me, revealing details I had never noticed with the Oppo, though after hearing it on the Soul I was able to hear it on the Oppo.
  • Bass: Oppo has more power in the bottom octave (< 30 Hz). Soul is more controlled with better defined bass timbre and slightly more mid-bass energy.
  • Transient response: Oppo has a bit more snap which sounds faster, but it also has a bit more ring / longer decay. Soul is cleaner, which can sound a bit “dead” at first but on deeper listen it doesn’t seem slower or smeared.
    • To avoid confusion, I didn’t try the Soul’s alternative minimum phase AA filter (though I’ve tried these before on other devices; the difference is subtle, but I usually prefer the linear phase “sharp” filter).
    • I did measure the effect of the Soul’s alternative AA filter. Comparing square waves, it eliminates pre-ripple, at the expense of rippling longer & louder after the impulse.
  • Dynamics: Soul is punchier with bigger macro-dynamics. Both have excellent micro-dynamics, though the Soul sounds darker between plucks/smacks, which hints at faster decay, lower noise or distortion.
  • It took time listening to a variety of music to establish a preference.
    • Sometimes the Oppo’s earthy airiness added realism and refinement. Other times, it slightly veiled what the Soul made more clear.
    • Sometimes the Soul’s tonal purity made the Oppo sound veiled in comparison. Other times, this purity sounded sterile where the Oppo sounded organic.
  • With more listening across a wide variety of music I came to find the Soul more transparent and true to the source. What clinched it was piano and voice, which highlight the Soul’s clean, pure midrange.

Engineering: Soul

• The Soul has several engineering features that differentiate it from other high quality solid state DACs:
  • Volume control: It changes metal film resistors in the gain-feedback loop, rather than attenuating a fixed gain, so there is no potentiometer in the signal path.
    • Advantage: lower noise and perfect channel balance at all volume settings, no loss of SNR at low to medium volume settings.
    • Sometimes with a stepped attenuator the perfect volume you want is between clicks. But this never happened with the Soul; it averages about 0.5 dB per click which is fine enough to set the perfect level.
  • Meier “FF”: The Soul’s digital and analog stages are frequency-shaped to reduce distortion and noise in the midrange and treble where the ear is most sensitive.
    • Jan calls this feature “FF” and describes it here.
    • That article is long and can be hard to understand. My simple take on it is here.
  • Power supply: The Soul has 4 switched power supplies with about 70 mF (a lot!) of filter capacitance: 1 for the digital section, 1 for the USB section, and 1 each for the positive and the negative supply lines of the analog stage. This provides near perfect DC with incredibly low noise and not even a hint of 50/60 Hz ripple.
  • DAC implementation: The Soul uses the Wolfson WM8741 DAC in mono mode (where it has a slightly higher SNR), one per channel (L and R). This chip’s analog output pins are balanced, which the Soul maintains all the way to its analog outputs. It also operates the DAC chip in maximum oversampling mode and enables the user to select which digital filter to use (sharp vs. slow).
    • Note: the Oppo uses the ESS9018 which has the ESS Hump, an anomaly that increases distortion at the low to medium levels used by most music.
  • I believe the above engineering features make the Soul sound subtly different from other top quality solid state DACs, and are the primary contributing factors behind my subjective listening observations.
• These features give the Soul a higher level of attention to engineering detail. From an engineering perspective, it’s the right thing to do if you want the best sound at any cost. As an engineer myself I believe in these kinds of features.
• Yet a music lover asks: does this get me closer to the music leading to greater appreciation and enjoyment? Possibly… yet in general not necessarily. With the Soul, I think it does.
• For example:
  • Years ago I built a stepped attenuator to replace my preamp. It sounded better than any active preamp I had heard. It revealed subtle musical details that even this very fine preamp (Rotel RC-990BX) veiled.
  • I enjoyed it for over 10 years until I replaced it with a dedicated DAC (the Oppo), which incrementally increased transparency.
  • Back then, the difference between my preamp and the attenuator were of a similar nature to what I heard from the Oppo to the Soul: incrementally improved purity and clarity.
• At this level of engineering and quality the equipment measures as perfect as engineering can make it. Reliably hear-able sonic differences may (or may not!) exist, but if they do, they are subtle and which is “best” is subjective.

Functionality: Tie (different trade-offs)

• The Soul has more DSP features: adjustable filters, EQ, channel mixing, etc.
  • I already have a digital parametric EQ (DEQ2496) supporting any number of bands. With this I can fine-tune the output more precisely than the Soul’s tone controls.
  • However, that fine-tuning comes at the cost of complexity: I spent hours carefully crafting each set of EQ with measurements and listening, then saved it as a named setting.
  • If I’m listening to the occasional music that is imperfectly mastered, the DEQ2496 is too cumbersome to EQ it on the spot.
  • The Soul’s controls are much simpler: 4 tone controls equally spaced at 2-octave intervals, digitally implemented.
    • Note: the tone control spacing is not equally spaced from the factory, but Jan Meier customized them for me with corner frequencies at 80, 320, 1250, and 5000.
  • No recording is perfect and I normally listen to how it naturally sounds, however imperfect. Yet some are more than imperfect, flawed to the point of distracting from the music.
  • Here, I use the Soul’s controls to apply a mild correction to get past the imperfections and closer to the music.
  • This also applies with headphone listening. The Soul’s cross-feed gives a nice correction to music sources that have artificial hard L-R stereo separation.
• The Oppo has more types of inputs and outputs, both digital and analog.
  • The Oppo has Bluetooth and handles a wider range of digital formats (DSD, and additional PCM sampling frequencies).
  • The Soul doesn’t have unbalanced inputs or outputs, so you’ll need an unbalanced → balanced converter for unbalanced RCA audio sources.
    • In my case that’s OK because none of my unbalanced sources are reference quality (game box, computer).
  • With the Soul you’ll need balanced cables for your headphones and if you use its line-outs you’ll need XLR cables for your power amp.
• USB
  • My Android phone or tablet never worked with the Oppo’s USB input (it was made for Apple devices).
  • But, they do work with the Soul, and apps like USB Player Pro stream the bits without modification, so a mobile device becomes a fully transparent audio source.

Build Quality, Durability, Support: Soul

• Both have great build quality.
• Both get warm during use, but the Oppo much warmer than the Soul–possible longevity disadvantage? The Oppo volume control has a reputation for failing.
• Support: Meier sets an example for the trade with his engineering expertise and enthusiasm for music and engineering. He is responsive and direct with questions and feedback. I’ve never seen better support.
• The Oppo is built better than most consumer gear, both internal (big toroidal power supply, high quality opamps, etc.) and external (case, knobs, etc.).
• But the Soul has the edge here as it levels up to professional hand-selected parts and is built by Lake People in Germany.
• I’ve owned Meier’s Corda Jazz for several years of daily use. It shows no signs of wear; the switches, knobs, case, etc. all like new. It’s at least as solidly built as the Oppo, and the Soul is a step up from there.
• Ten years from now, which is more likely to still be running like new? Probably both, but if I had to pick one or the other, no question it’s the Soul.

This has been a fun and educational week, though my ears and brain will take time to recover from all the critical listening. Good consumer gear has gotten very good indeed, raising the bar. From objective measurements alone, it can be indistinguishable from the best of the best. Yet even someone with an “engineering-first” attitude (myself included) must admit that even gear whose measurements show all forms of distortion below theoretically audible thresholds, still can sound different. We measure much of what we hear, even most of what we hear, but we don’t necessarily measure everything we hear, and the quirks of perception acuity can sometimes surprise us.

The Oppo HA-1 is no longer made, so it’s hard to recommend despite being a fine piece of kit. But if you can find one on eBay, it’s hard to find its equal in sound quality under a kilobuck, and it’s super flexible having many inputs and outputs. However, if you want a DAC, line stage and headphone amp that is among the best available at any price, I recommend contacting Jan Meier and listening to the Soul. Sadly, some expensive high-end gear is just audiophile bullshit. The high price is mainly about fancy cases and knobs, low production numbers, and social signalling exclusivity. It’s great to see engineers like Meier bust that stereotype, justify the price with real engineering features, and demonstrate that well engineered and built equipment really can sound better (even if only slightly, since the bar is so high) and get us closer to the music.