Category Archives: Audio

Audio Phase: Shift versus Inversion

It is said that a 180* phase shift is the same as a polarity inversion. That is, it flips a wave to its mirror-image across the time axis. If we imagine a simple sin or cos wave, we see that this is true. 180* is half a wavelength, slide it that distance either forward or back, and you get the same wave with polarity inverted. Another consequence of this lies in audio room tuning. If the distance between 2 walls is half a wavelength of a particular frequency, the wave reflecting from the wall, being inverted polarity, cancels the wave arriving which causes a dip or null at that frequency. Those same walls will also boost waves at twice that frequency because that same distance between the walls is their full wavelength, so the reflected wave is in phase with the arriving one.

But this doesn’t work with a general musical waveform. No amount of sliding it left (back) or right (forward) in time will invert its polarity. Intuitively, we see that a musical wave is not symmetric or repeating like a sin or cos wave. The musical waveform is much more complex, containing hundreds of frequencies all superimposed. Any distance we slide it left or right represents a phase shift at only 1 particular frequency. Alternately, sliding it left or right can be seen as a phase shift at all frequencies, but a different phase angle for each, since the distance it shifted is a different number of wavelengths for each frequency it contains. As in the above example, it boosts some frequencies and cuts others. This is what happens in a comb filter.

Since every frequency has a different wavelength, it’s hard to imagine how a phase shift of the same angle at all frequencies could even be possible. It is possible, but to do it we need to expand into another dimension and use complex numbers. That computation creates a new waveform that is the polarity inverted version of the original. You can find explanations of this all over the internet, for example here: https://www.audiosciencereview.com/forum/index.php?threads/analytical-analysis-polarity-vs-phase.29331/

Because of this, when speaking of music and audio I prefer the term “polarity inversion” to “180* phase shift”. Even though they can be equivalent, the former is concise while the latter is somewhat ambiguous since one must also specify at what frequencies the phase shift is applied.

Tascam DA3000

The Tascam DA-3000 is a professional rack mount digital recorder. For years I owned a prior model, the SS-R1. It provided years of reliable service and I used it to archive nearly 1,000 vinyl LPs. The DA-3000 improves upon the SS-R1 in a few important ways:

  • Better AD converters: Burr Brown PCM4202
  • Better DA converters: Burr Brown PCM1795
  • Supports higher data rates: 24 bit, up to 192 kHz PCM and DSD 128 (5.6 MHz)
  • Direct AD-DA mode
  • Lower distortion and noise

It also retains many of the most important features of the SS-R1

  • Records to SD and CF media (no internal hard drive)
  • No fans – dead silent
  • The flexibility of many connections
    • Analog unbalanced RCA input and output
    • Analog balanced XLR input and output
    • SPDIF coax input and output
    • AES/EBU input and output
    • Internal or external clocks
  • Professional build quality, durability and reliability

Tascam no longer makes the DA-3000 so I bought mine used from eBay. In using it I’ve discovered some interesting quirks.

Date/Time Clock

The SS-R1 had a clock that had to be set every time you powered it on. Even when plugged in, it would forget the date & time when turned off. The DA-3000 fixed this – at least supposedly. But every time I powered mine up, I had to set the clock.

The problem was that the DA-3000 uses a rechargeable button cell battery to remember the clock when turned off. And it uses a tiny one that is soldered to the board. After a year or two, this battery dies and is not easy to replace (you must disassemble the unit, remove the board and solder). I contacted Tascam support and they said they no longer provide this service. It was annoying enough I decided to do my own permanent fix, better than what they would do at the factory.

Rather than simply replace that soldered-in battery, I installed a small battery cage for a CR2032 battery, which has the same voltage but is more than twice the size and capacity. Instead of soldering a new battery onto the board, I soldered the battery cage lead wires. Here’s what it looks like now:

I put an ML2032 battery into the cage (the rechargeable version of a CR2032). Not only will it last much longer than the tiny OEM battery, when it dies I can now replace it in 10 minutes easily without any soldering. This is how the DA-3000 should have been built from the factory.

DA-AD Mode

This mode stands for “Digital-Analog, Analog-Digital”. In this mode, the DA-3000 doesn’t record, but merely activates its DA and AD converters. You select which analog & digital inputs & outputs it uses. The DA converter has a slightly warm, soft voicing free of glare. Very nice. I can find no explanation for this in measurements, as its frequency response and distortion measure clean.

DA-AD mode does not auto-detect the sample rate. You must select the sample rate in the menus. If the sample rate you select does not match the digital input, the DA-3000 will still produce analog output but it is distorted. By this I mean high frequencies rolled off with elevated harmonic distortion.

Frequency response in DA-AD mode with sample rate mismatch, compared to sample rate match.

Distortion in DA-AD mode with sample rate mismatch:

For comparison, here’s distortion when you manually set the rate to match the input:

This distortion is measured after the DA3000 DA conversion and the SSR1 AD conversions. The distortion when you don’t manually set the sample rate to match the input is not documented in the manual, and Tascam support did not respond to my inquiry about this. So just know about this and set it!

Sample Rate Sensitivity

The popular S/PDIF digital format (whether coax or toslink) is a “push” protocol. The source device sends data to the downstream target using the source’s clock. The target device has no way to tell the source to slow down or speed up. No two clocks ever agree exactly, so the target device has to adapt to the source device sample rate. In contrast, audio over USB is a “pull” protocol. The downstream target device runs off its own clock, requesting data from the source as needed. No need to synchronize clocks or adapt sample rates.

Lest anyone disregard S/PDIF clock sync as a problem “solved in the real world”, consider that well engineered DACs use TCXO for their clock, which are temperature compensated crystal oscillators. These are typically accurate to roughly 1 ppm, which at CD quality 44.1 kHz makes a clock drift of 1 sample every 22.6 seconds. So the issue is real – with S/PDIF, the downstream device must adapt to the upstream clock. Buffering can’t entirely solve this, because it can only solve variations around an identical center frequency. Put differently, no two devices will ever agree on the center frequency, one will always be slightly slower than the other, which means any buffer you use will eventually under/over flow. With S/PDIF the downstream device must not only buffer the data, but also adapt its clock to the source rate.

The Tascam DA-3000 specs say it can sync to any input digital sample rate within a range +/- 100 ppm. This should be plenty, about 100x greater than the drift expected from a well engineered DAC. However, in my setup I have a toslink-coax converter between my Corda Soul (preamp/DAC) and the DA-3000. This converter causes quite a bit of jitter, so much that the DA-3000 occasionally loses sync. For example, a REW frequency sweep played through the Soul and captured on the DA-3000 looks like this:

That capture was at 88.2 kHz sampling, but it happens at all sample rates. My Topping E70 DAC handles this jitter just fine and is super clean, because it has a setting called DPLL that controls how much sample rate variance it can accept and adapt to. I had to bump it up a couple of notches to handle the switchbox.

Fortunately, the DA-3000 can do the same thing, even though one method for doing it is undocumented. When recording, enable SRC which is Sample Rate Conversion. This won’t actually convert the sample rate, because you’ll still manually set the DA-3000 sample rate to match the digital input. But when SRC is enabled, the DA-3000 accommodates and adapts to a wider range of jitter.

A better, cleaner method is to change the DA3000 clock setting from “Internal” to “DIN”. This tells the DA3000 to use the digital input as its own reference clock.

When you do either of these, distortion in the REW sweep is super clean like it should be:

AD Converter HF Noise

Another quirk of the DA-3000 is super high frequency noise in its AD converter. The noise is at 100 kHz, so you won’t see it at sample rates of 96k and lower since it’s above the Nyquist frequency. But at 176.4 and 192 kHz, it is there and looks like this:

So, if you are recording from analog inputs, don’t use 176.4 or 192 kHz. Use 88.2 or 96 kHz instead. The sound quality will be better! This is not an issue if you are recording digital inputs – that is just a bit perfect copy.

Analog Output Level

Normally, a recorder’s analog output level isn’t that important. But when using the DA-3000 in AD-DA mode, it becomes so. You need to match the input voltage sensitivity of the downstream preamp. The DA-3000 has a setting called “Reference Level” that sets this. Indirectly, this sets the analog output voltage for digital peak levels. The range is -9 dB which is +15 dBu (quiet) to -20 dB which is +24 dBu (loud). In Volts RMS this ranges from 4.36 to 12.28. The first is the consumer audio standard, the second is the professional standard.

Most consumer preamps have voltage sensitivity for their balanced analog inputs that expects peak levels around 4 V. Higher voltages can cause them to clip or distort. So you would set the DA3000 to -9 dB.

The Corda Soul has an internal switch to set its analog input voltage sensitivity. The default setting is for professional audio, expecting 12 Volts (low gain). Flick the switch to the other position and it changes to consumer, expecting 4 Volts (high gain). With the Soul set to low gain, the DA3000 setting that matches the output level of the Soul’s internal DAC is -16 to -18: -16 is about 0.5 dB quieter and -18 is about 1.5 dB louder.

Conclusion

The Tascam DA-3000 is a wonderful recorder. It is incredibly flexible, easy to use, with SOTA transparent sound quality and professional build quality. It has many other features not described here, since I’ve focused on its quirks. I’ve wanted one for years and I am so happy I finally found it!

Like any piece of gear, it has a few quirks as seen above. But none of these are serious problems, they all have workarounds.

Topping E70 DAC Review

Introduction

After I had to return my 2nd piece of Chi-Fi equipment due to poor build quality and support, I said I was done with Chi-Fi and would stick to other manufacturers. However, after Amir reviewed the Topping E70 on ASR, I couldn’t resist. A fully balanced DAC among the best and cleanest he has ever measured, for $350 would be too good to be true.

My Corda Soul is a DAC, preamp, headphone amp, and DSP processor. Of these functions, most are SOTA quality except for its DAC. It uses dual WM8741 chips which were great for 2007, but DAC technology has improved.

The topping E70 is a line level DAC and nothing more. That is:

  • DAC using ESS 9028 Pro chips
  • Analog outputs: both balanced/XLR and single ended/RCA
  • Inputs: SPDIF (coax and toslink), USB, and Bluetooth
  • Internal power supply
  • Digital volume control
  • Display showing sample rate and output level
  • High build quality with a metal case
  • Excellent measured performance, among the best at ASR

Setup

I measured the E70 and the Soul using my Tascam DA3000, which has excellent DA and AD converters. Better than my Juli@ sound card, but not as good as an APx555. This would later lead to a surprise due to misleading measurements…

The Corda Soul distortion profile has always looked like this (Room EQ Wizard Sweep at 48 kHz):

You can see above that noise is excellent (too low to be measured) and distortion is generally good at -100 dB, but 3rd harmonic is not so good, rising to -70 dB in the upper mids.

I never knew where exactly this 3H hump came from: the Soul’s digital stage, analog stage, or DA conversion. I recently discovered that it comes from DA conversion. More on that here. I set up the Soul to use the E70 as an external DA converter (it can do that!) and here is how it measured:

So ends the story, right? That’s what I thought, until I measured it at 192 kHz.

High Frequency Noise

The Soul always had a noisier sweep at 192 kHz, like this:

Each of those distortion plots peaks at the same frequency:

  • 2H (red) at 45 kH, and 2 * 45 = 90
  • 3H (orange) at 30 kH, and 3 * 30 = 90
  • 4H (yellow) at 22.5 kHz, and 4 * 22.5  = 90
  • etc.

So the plot is misleading. What’s actually happening is that there is HF noise at 90 khz, and the plot is interpreting it as if it were harmonics of lower frequencies – in other words, harmonic distortion. Thus, I assumed this was due to high frequency noise from its switching power supplies not being properly suppressed.

Here’s a different perspective on that same plot, plotting harmonics at their native frequency, which makes the above interpretation obvious:

But, the Soul’s power supplies (Meanwell IRM-20-24) switch at 65 kHz, not at 90 kHz. So where was that noise coming from? Maybe it wasn’t coming from the Soul at all.

Process of Elimination

I connected the E70 balanced analog outputs directly to the Tascam DA3000 inputs and recorded a 192 kHz sweep. Here’s what I got:

Or, looking at this plot the other way:

So that HF noise wasn’t coming from the Soul.

Next, I measured the Soul at 192 kHz using the E70 as its external DAC:

It’s essentially the same as the E70 direct, same shape just a few dB higher as it’s passing through an additional analog stage.

Setup – Conclusion

What I learned is that HF noise around 90 – 100 kHz is created by the Tascam recorder. When you record from its analog inputs, the signal passes through its A/D converters which introduce this noise, probably from a system clock or switching power supply. This noise is only at sample rates of 176.4 kHz and higher, because at lower rates, it’s above the Nyquist frequency so the digital filters kill it.

The E70 DAC is clean at all sample rates including 192 kHz. And the Soul’s analog stage is also clean. The Soul’s internal DA converters are not as clean, adding at bit of distortion at midrange-treble frequencies in the audible band. Thus, the Topping E70 addresses the Corda Soul’s relative weak point, and the combination gives truly SOTA audio reproduction.

One might ask why not simply use the E70 to directly drive my power amp? Why use the Soul at all?

  • The Soul serves as a convenient preamp, having multiple digital inputs.
  • The Soul has unique and valuable DSP functions: tone controls, headphone crossfeed, etc.
  • The Soul’s volume control is an analog stepped attenuator gain switch that is mechanically reliable and ultra clean with perfect channel balance at all settings.
  • The E70’s volume control is software which can glitch or lose memory, causing instant power spikes damaging speakers.
  • The Soul’s analog stage is clean and transparent, so there is no downside to its value and convenience.
  • Reliability and durability: the Soul is built like a rock by Lake People in Germany, well beyond Chi-fi quality standards.
  • Redundancy: if the E70 (or any other external DAC that I use) ever dies, I can use the Soul as a complete system while getting the DAC repaired or replaced.

E70 Review

Setup issues resolved, let’s return to the E70 review. It’s a simple black box with minimal controls:

  • Power switch on the rear left side
  • Rotary knob / button on the front right side
  • Capacitive touch button on the front left side
Features and Observations

You can leave the power switch on all the time and it will automatically power up and down as it detects a digital input signal.

You can set its output level to 4V (standard unbalanced) or 5V, which is about 2 dB louder with consequently higher SNR. 5 Volts may be too much for some devices, so make sure it is compatible. However, if this is the case, I recommend using 5 V anyway and setting the digital volume to -2 or -3 dB for reasons explained below.

I wish its high voltage output supported 16 V which is standard for professional balanced audio (as the Tascam DA3000 has). But it doesn’t.

It has digital volume control set in software. It might seem there is no reason to set it to anything other than max (0 dB). Yet setting it to -2 or -3 dB should give more headroom to better decode digital audio that is recorded too hot with intersample overs or clipping.

When the digital volume control is enabled, the display shows only the volume level, not the current sample rate. When disabled (always output max volume), the display always shows the current sample rate. I wish this were configurable. I’d like to enable digital volume and see the current sample rate.

The display can be configured to go dark and light up only when the knob is used. This is a nice feature yet it has a little bug described below.

The E70 has 7 different digital filters. Here’s how they measured at 44.1 kHz:

Filter #3 is the default, which is a decent choice. But it is minimum phase, so I switched to filter #1 which has the same response and is linear phase.

Drawbacks & Limitations

Whenever the input sample rate changes, the E70 emits a “click” / “chirp” to the analog outputs. Take care to adjust volume when changing recordings.

If while the display is dark you turn the knob to adjust the volume, the first knob click that wakes up the display changes the volume yet the display shows the prior volume before it changed, so the displayed volume is incorrect.

Firmware 1.04 adds the capability to set SPDIF sample rate lock sensitivity with a new setting called DPLL. This was essential for me because the default DPLL setting 5 didn’t handle 88.2 and 176.4 sampling well, so I upped it to 7 and it became clean. Setting 6 also worked but I figured I’d give it one extra nudge just to be sure. But Topping hasn’t yet put this firmware up on their support site. You can get an unofficial copy at ASR: https://www.audiosciencereview.com/forum/index.php?threads/topping-e70-stereo-dac-review.39188/post-1411763

Sound Quality (Subjective)

When level matched, the E70 is virtually indistinguishable from the Corda Soul’s built-in WM8741 DACs. There is no difference in voicing or frequency response. Yet when playing certain kinds of music there is a slight difference. The E70 better resolves layers of subtle detail in complex orchestral music. In recordings with moderate to heavy reverb/echo, such as from a cathedral, where the music can get drowned or saturated with reverb, it resolves the musical line more clearly. These differences are very subtle, audible with only some kinds of music, and easy to over-state. Yet they can be heard.

In contrast, the Tascam DA3000 in DA-AD mode (DAC only) shows a greater difference. It is voiced slightly warmer than the Soul or E70.

Conclusion

The E70 provides truly SOTA sound quality both subjectively and in measurements. It’s not perfect, having some firmware bugs, and Topping is not known for good support. And the build quality seems good yet long term reliability is unknown. But for the price (about $350) it cannot be beat.

Audio: DACs and Revelations

Introduction

It’s commonly held among audiophiles who understand electronics that well engineered and built DACs are audibly transparent. This belief comes from properly conducted double blind tests with well trained listeners, performing ABX testing, learning about how DACs work and how to measure them, reading detailed DAC measurements made by others and performing those measurements themselves.

However, “well engineered and built” is a loaded phrase. Some DACs use a AA filters that start attenuating within the passband (below 20 kHz), or have passband ripple, or non-flat phase response. Some DACs have elevated IM distortion at moderate levels (the ESS IM hump). Others have increasing distortion near full scale. Some modulate power or clock noise into the outputs due to insufficient filtering. All of these limitations can be audible under the right conditions, and have been observed with DACs considered to be well engineered and built.

Also, “audibly transparent” is a loaded phrase. Does it mean musically transparent, or perceptually transparent? Consider the  difference between 96 kHz and 44.1 kHz rate, or a linear vs. minimum phase filter having equal amplitude responses. These differences are considered to be inaudible. I can differentiate them in an ABX test, but only using “appropriate source material”. In this case, a high quality recording of jangling keys or a square wave. Differentiating them with a musical signal is much more difficult, and I’m not sure I could do that. In some cases, I’ve detected these differences with high quality castanet recordings, but is that really music? I consider it on the borderline between music and test signal. We listen to music, not test signals, so while I believe a good audio system should strive for perceptual transparency, some people consider the lower bar of musical transparency to be sufficient.

The Corda Soul is a DAC, preamp, and headphone amp with useful DSP functions. I’ve owned one for nearly 5 years and in some ways it’s one of the best measuring pieces of gear I have seen. Subjectively it sounds fantastic (by which I mean it is transparent, or doesn’t sound like anything – and many DACs and preamps do not achieve this, adding their own coloration to the sound) and I prefer the Soul to other high quality DAC / preamps in direct comparisons. I never expected to encounter a better sounding DAC…

The Setup

I recently replaced my Tascam SS-R1 with the newer model, the DA3000. The SS-R1 still works like new but is limited to 44.1 k and 48 k sampling, where the DA3000 supports every sample rate from 44.1 k to 192 k and DSD 64 and 128. This means I can connect the Soul digital output to the DA3000 digital input, and the DA3000 will simply work regardless of the sample rate. The DA3000 analog output (balanced XLR) goes to the Soul’s analog input.

I tested the DA3000 and the Soul by playing test signals through the Soul and capturing its analog output on the DA3000. The Soul always had a small hump in the distortion curve that peaks at -70 to -80 dB at 1,000 to 2,000 Hz, and appears at every sample rate. Since I now had two recorders – both the DA3000 and the SS-R1 – I was able to narrow down what causes this by bypassing the Soul’s DA converters, using only its analog gain stage. I connected the Soul’s digital output to the DA3000, then sent the DA3000 analog output to the Soul’s input, then recorded the Soul’s analog output on the SS-R1. The hump disappeared entirely!

This means the Soul’s distortion hump was coming from its internal DA converters, or its FF de-emphasis curve which is implemented in DSP (both of which are bypassed when you use an external DAC). This is surprising, since the Soul goes to great lengths to ensure clean DA conversion. It uses well regulated switching power supplies and dual WM8741 DAC chips, each in mono mode, one for each channel, fully balanced. However, the Soul’s analog gain stage measured entirely transparent. Noise was below any threshold I could record. The SS-R1 is only 16 bit, so all I can say is that noise is below -96 dB even at low volume settings. Frequency response was perfectly flat. Distortion measured at -96 dB, the limits of 16-bit.

So: the Tascam DA3000 DA converters measured cleaner than the Soul (here for measurement details). And not slightly cleaner, but a whopping difference: from -70 dB to below -96 dB, at least 26 dB and probably more. And this is in a frequency range where our hearing is most sensitive. But that said, not every difference you can measure, is audible…

The Revelation

The Soul allows the use of an external DAC and has a switch to instantly switch between that and its internal DAC. The difference is readily audible, by which I mean I can hear it not only with test signals but on a wide variety of music. Perceptually and subjectively, compared to the DA3000, I characterize the Soul’s internal DAC as:

  • Slightly edgier, tonally as if adding just a smidge of upper midrange
  • A tad grainier, or less pure
  • Bass is a bit less prominent, but this could be subtle perceptual masking from slightly emphasized upper mids
  • Soundstage is a bit narrower
  • About 0.2 dB louder

In contrast, the DA3000 DAC sounds a touch more pure, more open, with more natural bass and a bigger soundstage.

I call this a revelation because it was so unexpected. It really surprised me. Up to now, the Soul has been less edgy / grainy than other DACs I have owned, such as the Oppo HA-1. Even though the difference is subtle, it is a joy to listen to my familiar recordings with a slightly smoother, more natural perspective.

Note: The 0.2 dB loudness difference is the obvious culprit. It’s small enough to be barely perceptible as loudness, yet perceived indirectly as “richer”, “more detailed”. Yet normally, all else equal, slightly louder is perceived as slightly better. So it’s the opposite of expected. And I hear the same subtle differences even after adjusting for the 0.2 dB loudness difference.

But Wait, There’s More!

Upon further listening I made some other observations. The Tascam DA3000 DAC doesn’t resolve fine detail quite as well as the Soul. It slightly veils some of the subtle background sounds. However, in voicing and soundstage I still preferred the Tascam. So the difference was more of a trade-off.

The Conclusion

The Soul is still a keeper. As an analog preamp, it is unmatched both subjectively and objectively: clean and transparent with noise and distortion so low I can’t measure it, and perfect channel balance at every volume setting. Its DSP functions are useful and well implemented. And it is well built with great support.

However, I am now looking at other DACs to potentially bypass the Soul’s internal DAC. More on this here.

If my opinion isn’t clear by now, I’ll just say it. Well engineered DACs do not all sound the same. Some may sound the same, while others may have audible differences. All audible differences can be measured – if you know what to measure and how to do it right. But most published specifications are only the most basic measurements that don’t cover everything that can be heard. So just because basic specs like SINAD and FR are the same, doesn’t necessarily imply they sound the same.

Schiit Jotunheim 2 Review: DAC+Preamp+Headphone Amp

Introduction

Note: about a year ago I got an SMSL SU-6 DAC. More on that here.

I’ve always enjoyed listening to music on headphones at work. As we are returning to the office, I want to have high quality audio listening. My Etymotic ER6 IEMs sound great, but (A) they isolate all other sounds, so when people walk by and say “hi” I don’t even hear them, and (B) they don’t reproduce the top half-octave, so while they do sound clean, there’s something subtly missing. I still have my old Sennheiser HD-580 which are still as good as new, but they have low voltage sensitivity so I needed an amp to drive them.

I wanted to play music from my phone (USB Audio Player Pro), my laptop, or my desktop. And most (but not all) my music is on a small external hard disk which occupies the phone USB port, so when playing from the phone I may use its USB output or its analog headphone jack. But when playing from the laptop or desktop, I’ll use USB since their built-in DACs are crappy and don’t handle sample rates above 48k.

So I needed an audio device that is a DAC with USB input, also analog input, with a built-in headphone amp. Furthermore, I have limited power plugs at work so I couldn’t use separate devices having external wall-wart power supplies. I needed this to be a single box with an internal power supply and standard power plug. And of course having excellent audio quality in its DAC and amp, with sufficient power to drive my Sennheiser HD-580. And after my recent experience with Topping and SMSL, made in USA with a good warranty and support. And not too expensive.

The Schiit Jotunheim 2 with DAC module is the only device that meets all of the above requirements, so I ordered one. The Asgard would also meet these requirements, so which to get? I opted for the Jotunheim because:

  • It has both single-ended and balanced outputs and inputs.
  • It has slightly cleaner audio (lower noise & distortion), and more power.
  • It has a better volume knob (Alps RK27 blue velvet) with better channel matching.
  • It has switch-selectable preamp outs.

Amir reviewed it at ASR a few years ago, when it had the prior version of the DAC card that wasn’t so great. He found it to be a great amp & preamp with a crappy DAC. Since then, Schiit revised and greatly improved the DAC. The DAC is a plug-in replaceable internal module/card that costs about $100, so folks who bought an earlier Jotunheim (or Asgard) can also upgrade to the new DAC.

Photos

With its all-metal construction, switches and knobs it has a look that says, “tools, not toys”.

Removing the cover reveals clean layout and construction, and that awesome Alps RK27 Blue Velvet volume potentiometer.

The rear view shows the flexibility of this one-box-does-it-all device:

 

Summary

The Jotunheim has

  • Internal power supply, no wall wart.
  • DAC with USB C input
  • 2 Analog inputs: RCA and XLR
  • 4 Analog outputs
    • Line level RCA
    • Line level XLR
    • Headphone balanced (4-pin)
    • Headphone unbalanced (1/4″)
  • Switchable gain: low and high
  • Switchable line outputs (the don’t auto-mute when headphones are plugged in)
  • Analog volume control (Alps RK27)
  • High power, low noise and distortion
  • High build quality (all metal construction, knobs, switches)
  • Made in USA with excellent warranty and support

The Jotunheim does not have

  • Digital display: does not show sample rate, bit depth, etc.
  • S/PDIF or Bluetooth digital inputs: it has USB C only
  • DSP algorithms: no tone controls, crossfeed, etc.
  • Perfect channel balance: the Alps RK27 is one of the best, but no potentiometer is perfect

Measurements

I measure using my desktop PC and Juli@ sound card with Room EQ Wizard software. The Juli@ sound card is not up to professional measuring equipment standards, but it is one of the best PC sound cards. My measurements are good enough to detect any flaws that might be audible, and some others below audibility.

I connected the Jotunheim to the PC (running Ubuntu 18) via USB, connected the Jotunheim’s analog RCA (single ended) outputs to the Juli@ inputs, disabled PulseAudio, and let Room EQ Wizard do its thing.

First I ran frequency sweeps. Each was -1 dB digital level at every common sampling rate: 44.1, 48, 88.2, 96 and 192. All were ruler flat. The most difficult is at 44.1k since the transition band is so narrow. Many DACs have ripple or roll off before 20 kHz. Here’s the Jotunheim:

It is down -0.1 dB at 14 Hz and 19,900 Hz. There is no ripple and the phase response is dead flat, which tells us it uses a linear phase digital filter. This is great for 44.1k sampling. The only drawback at 44.1k is that it doesn’t fully attenuate until 24.1 kHz, which is above Nyquist. This leaks HF noise, but it should be benign as all aliases must be > 20 kHz (inaudible). Higher sample rates are flat to much higher frequencies and fully attenuate by Nyquist. For example here is the Jotunheim at 192k Hz:

Here the low frequency roll-off and phase shift is in the Juli@ card (it also appears in loopback mode). The Jotunheim is down 0.1 dB at about 62 kHz. I would prefer to see a more gradual filter that uses the entire transition band (20k – 96k), but it doesn’t seem to suffer from this sharp attenuation.

Here is distortion & noise at 44.1k at max volume, low gain:

We’ve got something interesting going on here: surprisingly high 2nd harmonic (2H) distortion. It’s below 70 dB which should be inaudible, but could become audible for low level signals. For example if the music was at -30 dB, at 3 kHz where our hearing is most sensitive, this distortion is only 48 dB lower which could be audible to some people under the right circumstances.

Note: this measurement is not an anomaly. It matches Schitt's specs, which quote THD on single ended outputs at .03%, which is -70 dB. The Jotunheim is optimized for balanced outputs, which measure about 100x or 40 dB cleaner.

This smiley shaped 2H distortion appeared at every sample rate, at the same level. I suspect it comes from using the Jotunheim’s single ended RCA outputs. It’s optimized for the balanced outputs and if I read Schiit’s description correctly, it converts to single ended by ignoring the inverted polarity signal instead of differencing it. Differencing would eliminate 2H distortion. Some balanced circuits are so clean they don’t need to be differenced, but others require it.

I tested this theory by playing a frequency sweep from my phone using USB Audio Player Pro in bit perfect mode, connecting the phone’s USB output to the Jotunheim, and the Jotunheim’s balanced XLR line level outputs to my Tascam SS-R1 recorder. Here’s what I got:

Ah, this is more like it! Noise and distortion around -100 dB in the bass to -92 dB in the treble. This uses the Tascam SS-R1 recorder’s balanced analog input and A/D converter, so it’s truly excellent.

Gain and Output

The Jotunheim has two gain settings: low and high. Low gain at max volume is unity. High gain is 12.7 dB louder than low gain, or about 4.3x the voltage, which is about 18.5x the power. I find low gain more than sufficient even for my insensitive Sennheiser HD-580 headphones when playing from digital sources having -6 dB pre-attenuation.

The Jotunheim is a truly balanced, differentially signalled amp. The 1/4″ headphone jack output level is about 6 dB quieter than the balanced headphone jack.

Volume Knob

The next thing I measured was the volume knob. Analog potentiometers are a common weak point in any preamp or headphone amp. They never have perfect channel balance, especially at the lower knob settings which we use most often.

Here’s a frequency sweep using the Jotunheim’s single-ended RCA outputs, on low gain with the volume at the 12:00 position:

The smile shaped distortion curve is gone. Turning down the volume eliminated it. This appears related to the Jotunheim’s internal amp, which Schiit calls “continuity”. If I read Schiit’s description correctly, “continuity” means a class AB amp, but it’s biased high enough to operate in symmetric class A up to about 500 mW output (according to Schiit). I suspect that when you turn the volume down to 12:00 it’s below the output threshold, and symmetric class A (even though single ended) which eliminates that 2nd harmonic distortion. I didn’t expect a transition from class A to AB to make such a difference, and it could have a different cause.

Anyway, back to the volume knob channel balance. No knob is perfect, each individual knob is different, and remember this is an Alps RK27 Blue Velvet knob. If you aren’t impressed, consider that this single part alone costs a whopping $40!? I’m not kidding: here it is at Mouser.

OK so here’s a table showing each of the clock volume knob positions, attenuation and channel balance. Obviously, there’s a margin of error in positioning the knob, so the numbers are all approximate. I’ve added the JDS Atom volume knob for comparison, which I think uses an Alps RK09. That’s a good potentiometer, but a cut below the RK27. Also, the Atom 2 which uses a hand-matched Alps RK09. You can see that the Atom 2 is as good as the Jotunheim.

ClockJot LevelJot DiffAtom LevelAtom DiffA2 LevelA2 Diff
05:00 (max)N/AMatchN/AMatchN/AMatch
04:00-1.3Match-0.8Match0Match
03:00-3.6Match-2.1Match0Match
02:00-6.0Match-5.3Match-2.25Match
01:00-10Match-8.8Match-6.25Match
12:00 (half)-16Match-14.3Match-17Match
11:00-20Match-18.0Match-19.5Match
10:00-25Match-23.4L +0.5-22Match
09:00-36L +0.5-33.8L +1.5-26R +0.3
08:00-48L +1.0-41.3L +1.0-40R +1.0
07:00 (above min)-74L +2.3-68.3L -8.0-60R +4.0

In summary:

  • Volume knob channel balance is matched to 0.5 dB or better for the top 3/4 of its range, from 09:00 to max.
  • At 09:00, which is -36 dB, the L is 0.5 dB louder than the R
  • At 08:00, which is -48 dB, the L is 1.0 dB louder than the R
  • At 07:00 (lowest non-zero), which is -74 dB, the L is 2.3 dB louder than the R

This is as good or better than any potentiometer I have measured.

Analog Input from Phone Headphone Jack

One way I plan to use the Jotunheim is to play music from my phone out its analog headphone jack. This can go wrong in several ways, so I measured it. I played an REW frequency sweep on my phone, using USB Audio Player Pro, connected its headphone jack (at max volume) to the Jotunheim’s single ended RCA inputs, recorded on the Tascam SS-R1 then imported into Room EQ Wizard for analysis.

TLDR; it’s super clean and should provide excellent sound quality.

Jotunheim on low gain, max volume:

We can see it’s super clean, though the SNR suffers a bit due to the phone’s low max output level, it’s still 70-80 dB. The phone’s output level is so low, the Jotunheim at max volume doesn’t trigger its unusual smile-shaped distortion curve.

Jotunheim on high gain, max volume:

This is just as clean, even cleaner. How can high gain be cleaner than low gain? It’s not an equal comparison – the overall level is much higher/louder. The phone’s max output is so low that the Jotunheim on high gain max volume doesn’t overload the Tascam recorder inputs.

Distortion: Balanced vs. Single Ended

When I noticed elevated distortion from single ended line level RCA outputs at max volume, and discovered that it disappeared at half volume, I did some exploring to learn more about the relationship between knob position and distortion. Here are the graphs:

Once again, max volume (about 05:00 on the clock). The min at 200 Hz is about -96 dB, the peak at 10 kHz is -68 dB.

Here it is turned down just a bit to the 04:00 position: -92 @ 200, -70 @ 10k

Here it is at the 03:00 position: -91 @ 200, -72 @ 10k

Here it is at the 02:00 position: -90 @ 200, -75 @ 10k. The smile is flattening.

Here it is at the 01:00 position: -84 dB @ 200, -84 dB @ 10k. The smile is gone.

Of course we expect distortion & noise to rise relative to the signal as we turn down the volume. But how much? Let’s quantify this. At the 01:00 position, the signal is attenuated about 10 dB from max. The minimum distortion max volume is -96 dB; the minimum distortion at 01:00 is -84 dB. So when we reduce the volume by 10 dB, distortion drops by 12 dB. Since this involves eyeballing the position of the volume knob, there’s a margin for error so call it 1:1 linear. The distortion profile looks normal / flat up to about the 02:00 position, at which point a smile (rising distortion in low & high frequencies) just starts to emerge.

Again, this is only on single ended outputs. The balanced outputs are clean all the way up to max volume. At least as high as I could measure them – the voltage of the Jotunheim’s balanced outputs goes so high it overloads my sound card and Tascam. So I had turn the volume down in order to measure it. Summary:

  • Single ended/unbalanced output is as clean as balanced at low to moderate levels.
  • At high levels (02:00  on volume knob with full scale input), unbalanced output has slightly elevated 2H distortion, up to -70 dB in the mid-treble.
  • This elevated distortion should be inaudible in most cases.

Conclusion

The Schiit Jotunheim is a nice piece of gear. It does a lot in a single box, with an internal power supply (no wall wart). And it does it well, with good to great measurements. It also sounds great subjectively. It has high parts and build quality, metal not plastic, the volume knob is silky smooth with just the right amount of friction, the metal switches are a pleasure to operate, having the solid “smack” of professional equipment.

It works seamlessly from Ubuntu Linux, from Windows 10, and from my phone, with both digital USB and analog inputs. It didn’t reveal firmware bugs nor shut off during testing, like some DACs from Topping and SMSL have done. I didn’t encounter any issues recognizing it or sending music to it, nor any glitches on long-term playing. And I didn’t have to install any drivers.

The Jotunheim is inherently balanced and performs best in this mode with excellent near SOTA measurements. Yet it also has unbalanced inputs and outputs that measure good enough, and it supports all combinations across its inputs & outputs.

Finally, the Jotunheim is made in the USA with good warranty and support. It reminds me of the amps that Headroom in Montana used to build 25 years ago, only even better engineered and built, with more functionality. Years ago a device with this functionality, build quality and engineering would have cost thousands of dollars.

Audio: How Much Data is That?

It’s easy to compute but I figured I’d save it here for reference

RatebPSBPSKB/secMins/GBCD ratioNotes
44.1-161,411,200176,400172.271011.00Redbook CD
44.1-242,116,800264,600258.4671.50
48-161,536,000192,000187.5931.09
48-242,304,000288,000281.25621.63Standard DVD
88.2-244,233,600529,200516.80333.00
96-244,608,000576,000562.5313.27Popular for modern classical music recordings
176.4-248,467,2001,058,4001,033.616.96.00
192-249,216,0001,152,0001,125.015.56.53

This represents actual data bits to represent the music – no overhead. If you want to know what bandwidth is needed to carry an SPDIF signal at a given rate, add extra for packet overhead.

The formula is simple:

bits per second = S * C * B
S = sample rate (samples per second)
C = channels (2 for stereo)
B = bits per sample

For example for CD we have

S = 44100
C = 2
B = 16
S * C * B = 1,411,200 bits per second

Note: most DACs internally oversample before D-A conversion. They typically oversample at the highest integer multiple of the source rate that is less than their max rate. For example the Cirrus/Wolfson WM8741 has a max rate of 192k, so CD and DVD are oversampled 4x to 176.4 and 192 respectively. This happens automatically within the DAC chip. Because of this, it’s usually pointless to oversample an audio signal before feeding it to a DAC – the DAC is going to do it anyway, so why waste processing power and bandwidth doing it yourself?

Corda COUNTry: All in One Audio

Summary

The Meier Audio Corda COUNTry is a unique little device:

  • Equalizer, headphone crossfeed, reverb and other audio functions implemented in DSP (Digital Signal Processing)
  • Inputs (digital only): USB, SPDIF coax and toslink
  • Digital outputs: SPDIF coax and toslink
  • D to A converter
  • Analog output: single ended RCA (line level, volume controlled)
  • Analog output: single ended 1/4″ headphone jack

Its purpose is to adjust or tailor the sound to listener preferences. You can use it in many ways, though they fall into 3 categories:

  1. Pure DSP upstream from your DAC
  2. Headphone amp
  3. Preamp

In the first case you already have a headphone amp or preamp and you only want the DSP features. In cases (2) and (3) the COUNTry is an all-in-one device. The COUNTry was originally designed for (1) without any D/A or analog outputs. Later, the D/A and analog outputs were added since they increase flexibility with minimal cost impact.

The COUNTry operates internally at a single fixed sample rate: either 96k or 192k (buyer’s option). Digital inputs are resampled (if necessary) to this rate. I received a 192k unit.

The COUNtry

As auditioned, tested, measured

Features

I’ll review the features in the order in which I might use them most myself. With some of the features, like crossfeed and reverb, I couldn’t imagine a way to measure them. However, I did use and listen to all the features, and found that each does what it says and the effect is easily audible.

Measurements

I measured the COUNTry using a PC running Ubuntu 18, an ESI Juli@ sound card, with Room EQ Wizard software. I also have an SMSL SU-6 DAC and a JDS Atom amp. The Juli@ SPDIF coax output went directly to the COUNTry. I measured it in 2 ways:

  • COUNTry analog line level outputs to Juli@ analog inputs
  • COUNTry digital SPDIF coax output to SMSL SU-6 DAC, to Juli@ analog inputs

This measured the COUNTry as an all-in-one device with D/A conversion and analog output, and also as a DSP-only device upstream from a separate DAC.

The Juli@ sound card is high quality, but it’s nothing like professional measurement equipment from manufacturers like Audio Precision. I can measure the basics (frequency response and distortion), but take the measurements that follow as directional guidance within the limitations of my equipment.

Equalizer

The equalizer’s 7 bands have equal octave spacing with frequency ratio of 2.5:1 or 1.32 octaves. They are symmetric and complementary, so they can be combined without ripple. A picture’s worth 1000 words, so here are frequency response graphs of the positive and negative ranges respectively, showing each control individually, and all combinations.

The manual says each band is +/- 6 dB, which is roughly true yet oversimplified. The max effect of any single band is +/- 4.3 dB. Two adjacent bands maxed together is +/- 5.7 dB. Three or more is 6.0 dB.

This enables adjacent sliders to be combined to form new bands. For example, consider sliders 5 and 6, centered at 2.5 and 6.2 kHz respectively. What I need to adjust the LCD-2F headphones is a slightly wider band centered at 4 kHz. And that’s exactly what I get when using both together, as you can see in the above frequency response graph. And, you can use different levels of these 2 sliders to shift the combined center frequency a bit higher or lower, to get a center frequency anywhere between 2.5 and 6.2 kHz. This makes the COUNTry’s EQ even more versatile than it appears.

For example, the following chart shows two combos: band 4 at max with band 5 at half (in teal), and band 4 at half with band 5 at max (in magenta). You can see the center frequencies are 3 kHz and 5 kHz respectively.

Regarding slider sensitivity: the following shows the center 1 kHz band at positions 1/4, 1/2, 1, 2, 3, 4, 5 and 6:

The slider effects are nearly continuous down to a fraction of a dB, allowing very small gradations. Each slider has 6 marked notches on each side (positive & negative), though only the center position is a tactile notch. The first 4 marks are about 1 dB each. The 5th mark is another 0.7 dB, and the 6th / last / max mark is the same as the 5th.

I like this design: it works like a conventional equalizer, but is more flexible. It’s a creative solution to an old problem.

Bass Boost

What: a gentle bass lift starting at 200 Hz and gradually increasing to + 6 dB at 20 Hz. This closely follows the bass attenuation of the popular Sennheiser HD-580 and HD-600 headphones, which is nice. It also mirrors the COUNTry’s bass attenuation in high gain mode, so use it if you want flat response in high gain. Its measured curve is shown twice in the following graph:

  • The top red line versus the green line.
  • The purple line versus the bottom orange line.

The manual says it’s a shelf boost with center frequency of 45 Hz and Q=0.5, which is true to the measured response. Q=0.5 is such a gentle slope that it is +1 dB as high as 120 Hz (evident in the above graph).

Stereo Crossfeed / Expander

Some recordings have sounds hard-panned to the left or right channels. This sounds wonky on headphones. Unfortunately, this was a common recording technique in jazz albums from the 1950s and 1960s – for example the classic Miles Davis / Coltrane album Kind of Blue. Crossfeed mixes a little of these sounds into the opposite channel, with a brief delay; it narrows the stereo separation. When listening to albums like this, it makes it sound more natural. The COUNTry has 7 levels of crossfeed (the Soul has only 5), and an 8th switch position that is mono (standard mono without crossfeed or delay). I don’t use crossfeed for normal listening, but it’s one of my favorite features because it’s so nice for those albums that need it.

Note: when clicking into mono, the sound suddenly becomes more dull. This effect is purely perceptual – the measured frequency response is exactly the same as in normal stereo.

Stereo expander does the opposite, making stereo wider. It’s intended for speakers but you can hear the effect somewhat on headphones.

Notch Filter

Many headphones have a resonance that causes a spike in frequency response in the 6k to 11k range. The COUNTry has a notch filter that cancels these spikes. Here’s the measured response of all 15 of the settings. It measures true to the specified Q=2.0, -6 dB.

My Sennheiser HD-580 and Audeze LCD-2F don’t have such a spike, so I don’t use this feature. But the Sennheiser HD-800 does, as do many other popular headphones, making this a useful feature.

Reverb

This does exactly what it says: it adds reverb to the music. I’ve tried it with various recordings across a variety of musical styles from classical, to jazz, rock, etc. It’s not my cup of tea, as I prefer dry recordings. Yet this is a personal subjective thing. The reverb sounds natural and well implemented, and has 3 levels to play with.

Volume Control

The volume control feels analog but is digital, having perfect channel balance at every step from max to min. It also lights up an LED at certain exact positions: 0 (max), -6, -12, -18, and -24. This is useful for setting levels to avoid clipping when using DSP features.

Analog Gain

The COUNTry has 3 gain levels for its analog outputs: low, medium and high. Strangely, in high gain it attenuates low frequencies. The manual says this is to reduce excessive excursions of drivers at high levels. This doesn’t make sense to me, since high gain would be used for low sensitivity headphones or amps that need more voltage for the same sound level. And the COUNTry enables you to adjust the low frequencies if you want to. However, the bass boost switch exactly mirrors this bass attenuation, so if you are using high gain mode, turn on bass boost to get flat frequency response.

We’ve heard the mantra “the more gain, the more pain”. This is generally true, due to the constant gain * bandwidth of the transistors & opamps used in amplifiers. Because of this, the best approach is normally to use low gain whenever possible. That is, for a given loudness level, low gain at a higher volume position is usually cleaner than medium or high gain at a lower volume position. This also preserves digital resolution.

Note: This is especially true with conventional analog potentiometer volume knobs, which typically have better channel balance at high settings. However, the COUNTry's digital volume control has perfect balance at all settings. So that's not a factor here.

To test how the gain affects noise & distortion, I measured the COUNTry noise & distortion at each gain setting, with volume adjusted so that each had the same output level. Because the volume knob is digital, I couldn’t match the levels exactly, but I could match within 0.3 dB. More specifically:

  • Low gain at volume -6 dB
  • Medium gain at volume -16 dB
  • High gain at volume -26 dB

Medium gain measured slightly cleaner than high gain, but only slightly. Low gain was about the same as medium gain. So with the COUNTry, the gain level you use doesn’t make a significant difference in distortion or noise. Just select the gain that matches the sensitivity of the downstream device the COUNTry is driving. Of course, remember that high gain attenuates bass so use the bass boost to get flat response.

CD De-Emphasis

As mentioned above, the 7 band EQ has a creative design that eliminates ripple across the bands and enables adjacent bands to be combined to shift center frequencies. However, the highest band (16 kHz) has a special feature. When set to the minimum, it applies the Redbook CD de-emphasis curve. It’s nice to be able to apply this manually since some old CDs use it, they sound much too bright unless it’s applied, and many aren’t formatted properly for DACs to recognize and apply it automatically.

However, the equalizer slider has no indication when this triggers. It triggers at the bottom position, but as you raise the slider, it moves a short distance before disabling de-emphasis and attenuating the 16 kHz band. If you want the most 16 kHz attenuation without triggering de-emphasis, you must do it by ear. Slide it all the way to the bottom, then slowly lift it until the sound suddenly gets a bit brighter. It would be nice to have an indicator that lights up when de-emphasis is activated. The LED on the 16 kHz hand is already used as a power indicator, but it would be nice to have it change color or blink when de-emphasis is activated.

Miscellaneous

The measured impulse response is symmetric and phase response is flat into the high frequencies, which means the COUNTry uses standard linear phase filtering, rather than minimum phase.

The case and knobs are constructed in a manner that appears easy to disassemble and service. It’s good to see this exception to our modern age of cheap disposable equipment. I did not test this by opening the unit, since it is just a loaner.

Support is excellent. You can contact Jan Meier of Meier Audio directly on the company website or on Head-Fi. He is responsive, knowledgeable and helpful.

Usage and Build Quality

The COUNTry is a nice little device with an unusual look. It’s an “old-school” look and feel with physical knobs and LEDs instead of a display screen with menus. It has high build and parts quality.

Some of the indicators are a bit obscure but they become intuitive as you use them. For example, the digital sample rate display is done with 2 LEDs. One for the base rate (32, 44, 48) and another to indicate a multiplier (2x or 4x). Another LED lights up when the volume control is exactly at certain positions: 0 (max), -6 dB, -12 dB, -18 dB and -24 dB. This is a nice guide for setting precise levels.

The rotary switches make a solid “snap” and feel like professional test gear. The toggle switches are similar. The EQ sliders feel like potentiometers,  smooth with just the right amount of friction. The COUNTry is fully digital, so it detects the slider position and translates to a digital value. There is a tactile detent for the center position, but no others indicating when it’s shifted from one value to another. The controls feel good to use.

The headphone jack is seriously robust and overbuilt. The back panel connectors are not as robust but still feel solid. This makes sense as the headphone jack is likely to be plugged & unplugged much more often than the rear connectors.

Sound Quality (Subjective)

OK so what does it actually sound like? I listened on my Sennheiser HD-580 headphones. They are over 20 years old, but have fresh pads and I recently tested them with a MiniDSP EARS rig. They still have measured performance like new. I also used Audeze LCD-2F headphones (also tested & measured). The sound source was my PC with its ESI Juli@ sound card, and lossless FLAC files at bit rates ranging from 44.1 to 192. The Juli@ card coax SPDIF output routed to the COUNTry. I routed the COUNTry’s SPDIF digital output to my SMSL SU-6 DAC and JDS Atom headphone amp. With this setup, I could plug the headphones directly into the COUNTry, or into the JDS Atom. This enabled me to test the differences (if any) between the COUNTry in pure digital mode, versus its D/A and analog outputs.

I level matched by playing white noise, routing the analog outputs to my Juli@ card and measuring the level. With the JDS Atom’s analog volume knob, this enables level matching to about 0.1 dB.

In a word, the COUNTry sounds just a bit brighter than the SU-6 + JDS Atom. I first noticed this with music, and confirmed it with pink noise, which is a useful test signal for hearing subtle differences in frequency response. This brightness was unexpected, since the COUNTry’s analog frequency response has a slight bass boost and treble cut. However, the brightness sounds like it’s in the upper mids / lower treble, rather than in the highest frequencies. Loud parts with many instruments playing (like a symphonic crescendo) are slightly less resolved on the COUNTry analog output, compared to the SU-6 + JDS Atom.

Subjectively, the COUNTry analog output has a good clean sound across a variety of music with no obvious issues or artifacts. Though it’s not state of the art in terms of linearity, smoothness or detail – nor was it intended to be. Think of the COUNTry analog outputs as a sonically competent add-on feature. The COUNTry digital outputs are more transparent.

When you plug in headphones while the COUNTry is playing, it sometimes emits two loud CLICKs in the headphones, then pauses for about 1 second before the music starts. Inserting the headphone plug quickly usually avoids this. Meanwhile, the line level outputs are always playing and do not pause or CLICK. If you power on the COUNTry while it has musical input, you get a single loud CLICK then the sound starts playing in the headphones.

Anomalies

Frequency Response

Analog frequency response is not quite flat, and different in low (blue), medium (green) and high (red) gain. Medium gain has a 0.5 dB lift in low bass, and high gain has a 6 dB drop in low bass. All modes gently taper high frequencies, reaching about -0.6 dB at 20 kHz.

Note: in the above high gain curve (red), digital bass boost was applied to flatten the response. The uncorrected response is shown in another graph below.

Why the frequency response differences at different analog gain levels? This is due to the Meier Audio FF or Frequency Adaptive Feedback. It attenuates low frequencies in the digital inputs, then boosts them back to flat again in the analog outputs. This means the attenuation curve (implemented in DSP) must match the boost curve (implemented in passive analog components). Meier Audio optimized this matching at low gain, since they expect that will be most often used. The purpose of FF is to frequency shape and optimize DA conversion and the gain-feedback loop, so it is only used for the analog outputs, not on the digital outputs.

As mentioned above, in high gain the frequency response attenuates bass beginning at 200 Hz, gradually reaching -6 dB at 20 Hz (this is noted in the manual). The COUNTry has a DSP bass boost that applies the exact reverse of this. So if you want flat response in high gain mode, turn that on (as it is in the above graph).

Let’s take another look at the bass boost frequency response graph shown earlier. From bottom to top:

  • Orange is high gain
  • Purple is high gain with bass boost activated
  • Green is medium gain
  • Red is medium gain with bass boost activated

The COUNTry’s digital output is smoother and more linear, but still has gentle HF attenuation reaching -0.6 dB at 20 kHz, and even gentler LF attenuation about -0.2 dB at 20 Hz. We see below that this slight attenuation happens in the digital domain. This graph shows the digital output with all sampling rates on top of each other:

Noise at High Sample Rates

At high sample rates (176.4 and 192) the COUNTry has a lot of high frequency noise. This is unexpected and unusual, so I tested in several different ways:

  • Analog outputs
  • Analog outputs at lower levels (-6 dB, -12 dB)
  • Analog outputs with lower digital input levels (to -48 dB)
  • Analog outputs with input from my phone (USB Audio Player Pro) in bit perfect mode, instead of from my computer (ESI Juli@ sound card)
  • Digital outputs
  • Digital outputs at lower levels (to -48 dB)

It measured the same in all cases. This suggests the issue is not in the D/A conversion or analog stage, but in the DSP.

Here’s what that noise looks like at 176.4 kHz. Actually, in the graphs below it looks like distortion rather than noise, but more on that later. The horizontal line at y=0 dB is the sweep signal. It is not really at 0 dB, but this is relative to facilitate reading the levels. The brown line is noise. The black line is THD. The colored lines are 2H, 3H, etc., which sum to the black line.

At 176.4k, it’s clean up to 11 kHz and that first distortion spike is at 11,960 Hz. This is inaudible since it’s 7th harmonic so the distortion tone is at 11,960 * 7 = 83,720 Hz. However, these distortion spikes are big enough that their intermodulation (IM) differences are in the midrange. They are spaced 1-3 kHz apart, which puts the IM smack-dab where our hearing is most sensitive to it.

Here’s what it looks like at 192 kHz:

At 192k we have a similar situation, first distortion spike at 9,580 Hz, which will be inaudible since it’s 9th harmonic so the distortion tone is at 9,580 * 9 = 86,220 Hz. The 7th, 6th, and rest of harmonics are the same as at 176.4k sampling. And as above, the IM difference tones will be in the midrange.

Two interesting observations about these frequencies:

  1. They are exactly the same at 176.4 and 192 kHz. This suggests it is noise, not distortion – it’s not correlated with the signal.
  2. They all point to noise in the same region: 53 to 86 kHz

More precisely:

  • H9 @ 9,580 Hz means a frequency at 86,220 Hz
  • H8 @ 10,620 means 84,960
  • H7 @ 11,960 means 83,720
  • H6 @ 13,640 means 81,840
  • H5 @ 15,870 means 79,350
  • H4 @ 18,940 means 75,760
  • H3 @ 23,320 means 69,960
  • H2 @ 30,000 means 60,000

Here’s my speculation as to the root cause. Something in the COUNTry (perhaps a switching power supply?) is operating at 65 kHz. This is not fully suppressed or filtered and is leaking (somehow?) into the digital data. At sample rates 96k and lower, it’s above Nyquist and thus digitally filtered. But at 176.4 and 192, it’s below the Nyquist limit of 88.2 and 96 kHz respectively, so it is in the passband. Thus this power supply switching noise appears as harmonics of the given lower frequencies. If so, then it’s not really distortion, it’s noise.

Wherever the source of this noise, it’s interesting to study it closer. I created another graph below for this. Noise goes through the roof (literally – greater than 100% or louder than the fundamental) with spikes at 54.6 and 65 kHz. This noise profile and the spike frequencies are exactly the same at 176.4 and 192 kHz, so it’s independent of sampling frequency. It looks like this noise is what the graph (incorrectly) shows as distortion. Each of the N harmonic spikes is simply that same noise frequency divided by the next larger number (2nd, 3rd, 4th, etc.). Note that noise above the 0 dB line is not a bug. The spike at 65 kHz at +10 dB means it’s 10 dB louder than the signal.

In case this noise were caused by the high level of the digital signal (even though it wasn’t clipping), I ran another sweep at -48 dB. The noise profile looks the same:

In short, the noise is a constant level, and a constant frequency spectrum. It’s independent of the sample rate and of the input signal. This suggests it’s noise not distortion.

I said earlier that I was speculating – let me clarify. What is not speculation, but observation, is a lot of supersonic noise at high sample rates. My speculation is toward the root cause. I love a good puzzle and this explanation fits the observations. But I don’t really know what is causing that noise. So let’s set root cause speculation aside…

For comparison, here’s the COUNTry distortion and noise at 96 kHz, measured at the analog outputs at medium gain:

BTW, see that drop in noise & distortion from 700 to 2,000 Hz? I’m guessing this is the previously mentioned FF at work.

Here’s the distortion and noise of the measuring system, my ESI Juli@ sound card in loopback:

This typical of all sample rates 96 kHz and lower. So the COUNTry’s D/A and analog output is pretty clean – just a bit higher than the loopback connector, but not much. This should be well below audible levels.

Finally, I used Audacity to under-sample the 192 kHz test signal to 96 kHz, then played it through the COUNTry. The output looked clean, just like the 96k signal above. As expected.

Just in case the issue was with the 192 kHz tone itself, I measured how the SMSL SU-6 reproduces that test sweep signal, directly without the COUNTry. You can see those rising harmonics, but they cut off around -80 dB. This is the inherent noise and distortion of the Juli@ card’s analog input and A/D converter.

Conclusion

The COUNTry is a unique device. At 1 kilobuck it’s rather pricey, but considering its functionality, as an all-in-one device it replaces several others: DAC, amp and EQ with DSP. If you bought these separately, the total cost might be similar. And the COUNTry’s DSP features are unique and well implemented – you might not be able to find such a flexible EQ, or DSP with crossfeed or reverb. It’s built to last and the sound quality, both measured and subjective, is quite good if not state of the art. Used in purely digital mode, it should be sonically transparent, at least up to 96 kHz.

Meier Audio sent me one on request because I was curious about it. I don’t get to keep it. Yet over an extended trial using it nearly every day, here’s what I like best about it:

  • The flexibility of the EQ (shifting center freqs by combining adjacent bands) enables me to match the LCD-2F and HD-580 headphones, and most others.
  • The crossfeed is nice especially with albums having hard-panned L-R stereo separation.
  • Flexibility to be an all-in-one device or run in digital-only mode.
  • Works with my various audio systems with no observed software/firmware bugs.
  • Good quality, seems built to last with good support.

Suggestions for improving the COUNTry:

  • Fix the HF noise/distortion issue at high sampling rates.
  • For improved transparency, make a version that run DSP at the native sampling rate of the source without resampling.
  • If native rate processing is not possible, then over or under sample at integer multiples. Example:
    • Implement all internal functionality at 2 rates: 88.2 & 96
    • Input at 44.1, 88.2 and 176.4 process at 88.2
    • Input at 48, 96 and 192 process at 96
  • Provide an indicator when CD de-emphasis is activated (perhaps change the color of the 16k EQ band).
  • Eliminate the bass attenuation in high gain mode.
    • Sure you can normalize it with bass boost, but better to give it flat response like low & medium gain.
    • Not really an issue for me, since low gain is plenty for my headphones.
  • One of the nice things about the COUNTry is that due to the physical knobs you can see all the modes at a glance. But the knob detents are hard to see. If I owned this device I’d fill the detent notches with white paint.

Of the above, the the only thing holding me back from buying one for myself is the first 2 items. Eliminating the high frequency noise and not resampling might also improve the clarity / transparency / sound quality.

Meier Audio makes another version of the COUNTry that resamples everything to 96 kHz instead of 192 kHz. This 96k version has a more complex implementation of bass boost, called bass enhancement. It’s quite an interesting feature that leverages the psychoacoustic phenomena known as the “false fundamental”. As cool as that is, this 192k version has certain advantages, even if one doesn’t care about the difference in sample rate:

  • Bass boost exactly offsets the analog bass attenuation at high gain. The 96k version has no way to do this, since it replaces boost with enhancement.
  • Bass boost closely offsets the bass attenuation of several headphones, especially the Sennheiser HD 580/600/650/800 models.

Note that neither of the above reasons has anything to do with 96k versus 192k sample rates. In my opinion, 192k offers no real advantage over 96k because whatever advantages are gained from rates higher than the 44.1k CD standard, are fully realized at 96k.

Update: Apr 2023

Meier audio has a new version of the COUNTry:

  • Performs all DSP at the native sample rate (32, 44.1, 48, 64, 88.2, 96, 128, 176.4, 192)
    • Digital-only mode has no resampling, should be fully transparent
  • D/A conversion is always done at 96k, since the WM8716 chip is optimized for this rate
    • The analog audio output is resampled to 96k before D/A conversion
  • Equalizer step sizes are 0.5 dB resolution
  • Reverb replaced with 3 levels of bass boost (50 Hz shelf, Q=1.0) +2, +4 and +6
  • The original bass boost (45 Hz shelf, Q=0.5, +6 dB) is retained, and can be applied in addition.

Analog vs. Digital

From Merriam Webster:

Analog

  1. of, relating to, or being a mechanism or device in which information is represented by continuously variable physical quantities
  2. something that is similar or comparable to something else either in general or in some specific detail : something that is analogous to something else

Digital

  1. composed of data in the form of especially binary digits
  2. of, relating to, or using calculation by numerical methods or by discrete units

In terms of storing, transmitting and playing audio, each term is ambiguous, yet their different meanings are similar, which leads to confusion.

Analog

The key phrase with meaning 1 is “continuously variable”. A turntable needle tracking a record groove, a tape deck head responding to fluctuating magnetic fields on tape, are both continuously variable. Reflective and non-reflective spots on a CD are not continuously variable – it either reflects a laser beamed at it, or it doesn’t. A square wave transmitted along a wire is not continuously variable – it is either at its max voltage, or its min voltage, nothing in between.

However, if we look more closely at the last two examples, we realize that they really are continuous. A reflective spot on a CD doesn’t reflect back 100% of the light; it’s not perfectly smooth, some of the light is scattered and lost. Conversely, a non-reflective spot does reflect back some tiny amount of the light, even though it absorbs or scatters most of it. And while all the spots of the same type (reflective or not) are similar, they are not exactly the same; each is unique. A square wave does not switch from high to low, or low to high, instantaneously. That would require infinite rise time, which is impossible. And as it approaches the new voltage, it will overshoot or undershoot just a bit before it stabilizes. So the voltage actually does vary continuously from the high to low value, even if it spends 99.99% of its time at a high or low value. In this sense neither of these are as discrete as they first seem; they are almost but not quite discrete, but actually continuous.

In fact, the universe at the super-macro atomic scale at which we perceive and manipulate it, is continuous. It only becomes discrete at the subatomic/quantum level.

The other sense of “analog” is that it is an “analog” of, or actually resembles the thing it represents (closely related to the word “analogy”). A magnetic tape that encodes music, has a strong field where the music is loud and a weaker field where it is quiet. A turntable needle tracking a record groove physically moves over a bigger amplitude when the music is loud, smaller when it is quiet. The shape of the groove itself resembles the waveform of the music being played.

Music as we experience it, and as it passes through air as vibrations and pressure changes, is continuously varying. Analog storage of music fulfills both definitions of the term: it is continuously varying, and it physically resembles the music (in some way, directly or indirectly).

Digital

The first definition refers to binary digits. However, this does not fully capture the sense of what it really means. The rational numbers are continuously varying, in the sense that they are infinitely dense: between any two of them, no matter how close they are, lie infinitely many more. Mathematically, the rational numbers are not a true “continuum”, as they have holes – by holes I mean numbers that we know must exist since they are the solution to simple algebra problems, yet are not rational. For example, the square root of 2.

Yet pragmatically speaking, this is a distinction without a difference. It is impossible to detect the difference between rational and real numbers through observation or measurement of the physical world. For every irrational number R, for any small value ε, we can pick a rational number Q so that | R – Q | < ε. We can pick ε smaller than any means of physical observation or measurement. Indeed, ε can be smaller than relativistic uncertainty principles permit. So even in theory, not just in practice, it is impossible to discern the difference in the physical universe. The difference between rational and real numbers does exist, but it is a mathematical distinction, not a physical one.

So for purposes of analog vs. digital, the notion of “infinitely dense” is a sufficient interpretation of what “continuous” means. Numbers can be continuous. Of course, numbers can also be non-continuous or discrete: like the counting numbers.

Even binary digits can be continuous. Every rational number can be expressed in binary digits, though some of them require infinitely many binary digits. For example, 1/7 in binary is 0.001001001… but that is still a well defined and valid number. When people use the term “binary digit” they often mean stored in a computer. But binary is simply a numbering system. It can be, but doesn’t have to be, stored in a computer.

However, in a computer the manifestation (storage, transmission, computation) of numbers is necessarily finite. Thus these numbers cannot be “infinitely dense”, which means they cannot be continuous. They are discrete numbers – even floating points, because they have finite resolution.

Because numbers can be either continuous or discrete, they are a poor concept on which to base the definition of “digital”. So much for Webster’s definition 1; that colloquial usage leads to confusion.

A better concept is definition 2: that of being “discrete units”. Storing or encoding data as a set of discrete states. We often use computers and binary, but the number of states is immaterial: it can be 2 (binary), 3 (trinary) or whatever.

In short, a big part of the confusion around the term “digital” is this: Just because it uses numbers, doesn’t mean it must be discrete. And just because it’s discrete, doesn’t mean it must use numbers.

Digital audio is discrete, and it uses numbers. The first is essential, the second is an incidental convenience.

Once again: truly discrete phenomena do not exist in our universe at the super-macro atomic scale. Discrete means a set of states with nothing in between the states. This is easy to understand from an abstract logical perspective. But shifting between physical states cannot be instantaneous, because that would require an infinite rate of change, which requires infinite energy/power.

Manifestation vs Meaning

Put differently: what it is vs what it means

Put differently: the signal, versus the information

A signal is a phenomena in the physical universe that encodes information. The signal can be a radio wave, a telephone transmission, handwriting on paper, scratches on a clay tablet, etc. Signals contain or encode information. The sender translates or encodes the information into the signal. The receiver decodes the information from the signal.

All signals are continuous, by the simple fact that they exist in our universe which does not have discrete phenomena at the macro-atomic scale. But this does not imply that all signals are “analogues” of the information they encode. Some are, some are not. That depends on the encoding.

The same signal could have different meaning, depending on the encoding. If the sender and receiver do not agree on the encoding, they may believe the message has been successfully transmitted, when it has not. The receiver might decode the signal in a different way than the sender encoded, and thus receive a different message.

This illustrates the fact that the signal, and the message or information it contains, are two different things. One is physical, the other logical. Signals cannot be discrete, but messages/information can be either discrete or continuous.

We can encode discrete messages into continuous signals. For example: consider the discrete binary message 11111100110 (which happens to be 2022, the current year). We can encode this into a series of voltage pulses each of fixed duration, encoding each 1 as 1.0 V and each 0 as -1.0 V. The receiver can easily extract the 1s and 0s from the signal.

However, somebody who doesn’t know the encoding scheme may receive this signal and not even know whether it contains a message, let alone what that message is.

Advantage: Digital

So what is the big deal behind digital audio? Why back in the 1980s was it called “perfect sound forever”? It’s neither perfect nor forever, so where did that phrase come from?

When encoding information into a signal, and decoding it from a signal, discrete states have certain advantages over continuous varying. Consider the above example of positive and negative voltage pulses. The receiver doesn’t care exactly what the voltage is. He can interpret any positive voltage as a 1, and any negative voltage as a 0. If the signal gets distorted, like a bunch of ripple added to it, or the peaks vary between 0.7 and 1.3 instead of exactly 1.0, it won’t change the message. The receiver will still receive it perfectly intact. Of course, under extreme conditions the signal could be so distorted that the message is lost, but that takes a lot of distortion. Encoding information as discrete states is robust, and in most normal cases delivers perfectly error-free messages.

Now consider an analog encoding of information, like a turntable needle tracking a record groove or a tape deck head detecting the magnetic field on tape passing by. Here, the encoding is continuously variable. Every tiniest wiggle or variation has meaning, it is part of the message. Indeed, high quality audiophile equipment is designed to respond to even those smallest subtle signal changes. If the signal gets distorted, even slightly, the distortion becomes part of the message. This encoding is not robust; it’s much more difficult to tell the difference between the encoded message, and any distortion that signal may have suffered.

Summary

For precision, let’s use “discrete” instead of “digital” and “continuous” instead of “analog”.

Digital audio is information. The original music is a continuous phenomena, and is encoded into information as discrete states. Those discrete states are encoded into continuous form for physical storage and transmission, and can be decoded back into discrete states. We use this discrete encoding because it is more robust, relatively immune from imperfections and distortions in transmission and storage, which makes possible perfect transmission that is not possible with information encoded in continuous forms.

In short, digital audio is “digital”, but the means by which we store and transmit it is “analog”. We encode the digital audio information into analog form or signals, and decode or extract the digital information from the analog signals.

The theory and physics of discrete vs. continuous information and signals has been known since Claude Shannon and others developed information theory back in the early 20th century. The Shannon-Whittaker interpolation formula which is the basis for analog to digital to analog conversion, was known at least since the 1930s. So why didn’t digital audio exist until the 1980s? The reason is computing power – or lack thereof. The range and resolution of human hearing is high enough that it requires a lot of digital data to attain sonic transparency. We knew how to do it, but it took decades for computing technology to get fast enough to process the volume of digital data required.

Negative Feedback

Introduction

I’m not talking about people complaining about stuff.  This is about feedback in amplifiers. I first heard about this back in the 1980s when I got into this hobby, but I never understood it as well as I wanted to. I think this is true of many other audiophiles. Most of the textbooks and other literature discussing negative feedback is either so technical it’s hard for people without EE degrees to understand (I took some EE classes but my degree is in Math), or it’s so high level it amounts to hand-waving. Over time I’ve studied it more closely and gradually understood it better. I thought this intuitive description may give the fundamental understanding needed to then go off and read the more technical stuff.

Opamps and transistors (henceforth, “devices”) are used to amplify signals – in our case as audiophiles, music. But they do not have linear outputs. In fact, they are nowhere near linear, but grossly non-linear. This applies to their output relative to frequency, as well as to amplitude. Non-linearities are distortion, in one form or another

Negative Feedback: Definition

Negative feedback is a method to make these devices more linear.

The concept is simple: feed the device output back to its input, inverted. Some portion or % of the device output is inverted and fed back to the input. Hence the name: Negative means it is inverted, Feedback means it’s fed back to its input. Here’s a simple negative feedback circuit:

In the above diagram, Input (green) is the musical signal, which follows the arrows. A (blue) is the device. B (red) is a simple circuit that attenuates and inverts the output before feeding it back to the device input. The input to device (A) is the combination of the musical signal superimposed with an inverted portion of its own output.

You have now created a circuit or system whose response (Output) is the cumulative effect of the input signal, the device, and how the device responds to the combination of signal and negative feedback. This circuit has a linear response, even though the non-linear response of the device in the circuit has not changed.

Definitions: open loop is the device’s native response; closed loop is the response of the negative feedback circuit containing the device.

Open Loop Gain Feedback

Most devices have an open loop response that has (approximately) constant gain * bandwidth. This means in its output, amplitude * frequency is a constant. If frequency increases by a ratio of R, then amplitude (or gain) decreases by a ratio of R. This forms a linear slope dropping 20 dB per frequency decade. That’s because -20 dB is a ratio of 1:10, and a frequency decade is a ratio of 10:1, thus their product is constant (more on dB here).

For example, here is the gain-bandwidth curve for the AD797 opamp. It also shows phase which we aren’t discussing here, so I circled in red the gain-bandwidth curve. Note that it uses log scales on both axes (dB for Y, Hertz for X).

You can see that it drops almost exactly 20 dB per decade from 100 Hz to 1 MHz, then drops more steeply to 100 MHz. This is essentially the device’s native or “raw” frequency response.

This is a steep slope. If we take human hearing to be the range from 20 Hz to 20 kHz (not exact but a rough approximation), this is a ratio of 1000:1 which is 3 decades (10 * 10 * 10). Over this range, a device’s open loop response drops a whopping 60 dB! Intuitively, it amplifies low frequencies MUCH more than high frequencies. The AD797 is one of the cleanest best opamps for audio applications, yet clearly we must do something to correct its response before we can use it in an amplifier.

Non-Linearities

We said above that a device’s response is non-linear. A constant gain * bandwidth may be steeply sloped, but it is actually linear! So what part of the device is non-linear?

First, gain * bandwidth is not constant over the entire range of frequencies. Gain starts out flat, until reaching its first corner frequency where it begins to drop (below 100 Hz in the above graph, so not shown). This drop is not exactly linear, but only roughly so. Then it reaches its second corner frequency, where it drops faster (1 MHz in the above graph). Second, the device’s gain vs. amplitude is not linear either. That is, its gain ratio is not a constant – it responds differently to small versus large input signals. Both of these represent different kinds of distortion. Yet these examples are only 2 aspects of the more general concept of the device’s “transfer function“, which describes its output as a function of its input. The input to the transfer function is the musical signal. The output is the device’s response to that signal. The transfer function is not linear, not even close.

Finally, a device’s open loop gain is typically so large as to be unusable, like around 1 million to 1 (or 120 dB). For a usable amplifier, we need to reduce this gain down to roughly unity or 1:1. Sometimes less, sometimes more, but most audio applications are in the range of -30 to +30 dB. So we need something in the neighborhood of 100 dB of attenuation, or about 100,000:1.

This gives us 3 problems to solve before we can use a device as an audio amp:

  1. Flatten its steeply sloped frequency response.
  2. Reduce non-linearities in its transfer function (distortion).
  3. Reduce its unusably high gain ratio.

Negative feedback solves (or mitigates) all 3 of these problems. Essentially, it trades gain for linearity. In audio applications we have more open loop gain than we can use, which makes this an easy win-win trade.

Let’s take #3 first. Feeding the inverse of the device’s output back to its input obviously reduces gain. Indeed, the net effect would be zero, which is why we attenuate the negative feedback; we feed only a fraction of the output back to the input. It also makes the gain more stable. For example, suppose the device’s open loop gain changes with temperature. Negative feedback reflects any gain changes back to the input, offsetting them. If the device gain increased with temperature, the negative feedback gets more negative which shrinks the input accordingly, and vice versa). Closed loop gain no longer changes with temperature. Thus negative feedback reduces the device’s open loop gain (say, 1 million to 1) to whatever level you need (say, 1:1) and also makes it more stable.

Now consider #1: frequency response. If the device’s open loop gain-bandwidth curve looks like an inverted hockey stick, then its inverse looks like a right-side-up hockey stick. Feed this to the device input and the two curves cancel each other: you get a flat line. That is, constant gain versus frequency – otherwise called flat frequency response. This is true no matter what the shape of the device’s open loop gain-bandwidth curve – it doesn’t have to be linear.

Bandwidth is defined as the lowest frequency that sees a 3 dB drop in gain. If you have, say, 6 dB of negative feedback (half the output signal inverted and fed back to the input), the effect is like drawing a horizontal line across the device’s open loop gain-bandwidth chart, 6 dB  below the top. Response is now flat from 0 moving to the right until it intersects the original gain-bandwidth curve.

For example, here (below, red line) is what the above device (AD797) frequency response would look like with 60 dB of negative feedback:

The circuit now has 60 dB of gain, instead of 120 dB, with flat frequency response to 100 kHz. We’ve traded 60 dB of gain for flatter frequency response (wider bandwidth), plus other benefits described below. And that’s still more gain than we need in most audio applications.

More negative feedback means a horizontal line further down on the chart, which is flat to a higher frequency before it intersects the open loop gain-bandwidth curve and starts to drop. Knowing that the open loop gain-bandwidth curve drops 20 dB per frequency decade, we can quantify this effect. Every 20 dB of negative feedback gives you about 10x more bandwidth. This is just a rough rule of thumb because it depends on the shape of the device’s gain-bandwidth curve, which as mentioned above is not perfectly linear.

This effect of applying complementary frequency response curves is similar to the RIAA EQ curve for vinyl LPs. Before cutting the record, they apply an EQ so that 20 Hz is 40 dB quieter than 20 kHz (roughly linear across the spectrum). On playback, they apply the reverse EQ boosting 20 Hz 40 dB more than 20 kHz. The 2 curves cancel each other, giving flat response.

Step 1 complete: negative feedback flattens the frequency response and determines the bandwidth.

As you go further down the chart, you get more bandwidth and less gain. At some point you’ll stop. This point is determined by:

  • How much bandwidth you need
  • How much gain you need
  • How much gain & bandwidth you had to begin with (the device’s open loop gain-bandwidth curve)

Finally, consider step 2: distortion. This is merely a generalization of step 1. Negative feedback applies to the device’s transfer function; frequency response is only 1 aspect of that. By offsetting the transfer function, negative feedback corrects, reduces or squashes all non-linearities, for example distortion like harmonic and intermodulation.

In summary, we have explained how negative feedback achieves 3 goals:

  1. Flatten frequency response, widen bandwidth
  2. Reduce distortion
  3. Stabilize gain at a desired level

Timing & Phase

If negative feedback is delayed, then the negative waveform fed back to the input won’t line up with the device’s output. Thus, it can fail to offset the device’s transfer function, and may in some cases exaggerate it, turning into positive feedback. This is not normally a problem for audio, since audio frequencies are so low and slow compared to the device’s open loop bandwidth. However, if the gain-feedback loop has devices that alter the phase or have different impedance at different frequencies (capacitors or inductors), this could cause the same problem. Normally, the gain-feedback loop consists only of resistors to attenuate the signal to the desired level. In audio applications, metal film resistors are used due to their ideal noise profile.

In short, as long as the gain-feedback loop consists only of metal film or similar resistors, negative feedback should not introduce timing or phase issues in audio applications.

These Aren’t the Frequencies You’re Looking For

If you look at distortion specifications for amplifiers, you might notice that distortion usually rises with frequency. One reason for this is the shape of the device’s open loop gain-bandwidth curve. Its steep slope of -20 dB per decade means the device’s native output, and hence the negative feedback signal, consists mostly of low frequencies. This means low frequencies get more correction than high frequencies, which means lower distortion. Conversely, high frequencies have less feedback and consequently have higher distortion. Frequencies at the upper end of the bandwidth have almost no correction – they are so small in the device output, and thus also in the negative feedback.

This is the opposite of what we want because our hearing is much more sensitive to distortion in the middle to upper frequencies.

For example, highlighted in green in the diagram below, 60 dB of negative feedback as shown above gives us 60 dB of correction at 100 Hz, but only 15 dB of correction at 20 kHz. At 100 kHz where the red curve meets the black curve, the level of correction shrinks to zero.

Here we’re talking about audio frequencies — kilohertz, not megahertz. Most devices have bandwidth into the megahertz range. A common solution is to use more negative feedback than you “need”, to make the bandwidth wider than the audio spectrum. This puts the highest audio frequencies in the middle of the bandwidth so they have enough feedback to reduce distortion. Of course, relatively speaking, they still have less feedback than the bass frequencies.

SMSL SU-6 Review: DAC + Preamp

Introduction

I’ve said this before but it’s worth repeating: when it comes to audio, we are spoiled with an abundance of riches. DACs, amps, headphones have gotten so much better over the past 15 years, it’s hard to imagine that this was sometimes considered a “solved problem”. At the same time, prices have gone down, not up.

A few weeks ago I got a Topping EX5. It was a great little device but it had buggy software and Topping could not fix it. So I got an SMSL SU-6 to see if would be any better. Amir reviewed the SMSL SU-6 favorably at ASR. Refer there for detailed measurements.

Here I’ll fill in some of the details that Amir doesn’t cover: whether the SU-6 has the same software bugs as the EX5 (or others), and details on its digital filters.

Here’s what the SU-6 looks like in service – my computer desktop audio stack: the SMSL SU-6, the JDS Subjective 3 EQ, and the JDS Atom headphone amp. The slight bass boost compensates for the Sennheiser HD-580 bass rolloff.

Summary

TLDR: The SMSL SU-6 is simpler than the EX5. It is a single-ended, line-level DAC, only that and nothing more. No headphone amp, no balanced outputs. It does have a volume control. I bought it for $170 on Amazon. This is my second piece of “Chi-Fi” (Chinese Hi-Fi) equipment, the EX5 being the first.

Compared with other well engineered DACs having excellent measured performance, what distinguishes the SU-6?

  • Display properly shows current sample rate
  • Volume control, useful to turn it down a few dB to avoid clipping hot sources
  • Several digital filters from which to pick
  • Adjustable phase lock to minimize jitter and prevent audio glitches
  • Internal power supply – no wall wart

Update: Feb 2023

My SU-6 which has been reliable for just over a year, just developed a problem. I was listening yesterday and it didn’t sound quite right. Too subtle to pin down, so I measured it. It has developed THD at -55 to -60 dB, all 2nd harmonic. It used to measure clean (THD below 90 dB which is what my sound card measures in loopback mode). I got in touch with SMSL support via email and the outcome is pending. As I check reviews on Amazon, it seems the SU-6 has more than its fair share of failures and support can be difficult.

Resolution

Support was difficult to contact and led to a frustratingly slow drawn out email chain with people having limited English language skills. Finally they agreed I could send it back for repair, but I chose not to:

  • I have to pay to ship it to them for repair.
  • Mailing packages to China is prohibitively expensive.
  • There’s no guarantee it will arrive, could get held up or lost in customs.
  • If it does arrive, there’s no guarantee the company will actually fix it. The unit was 100% functional but had slightly distorted analog output. They might do a simple function check and return it as “nothing to fix”.
  • If they do actually fix or replace it, there’s no guarantee it will arrive back home. It could get held up or lost in customs.

For these reasons, I considered it a $190 lesson and opted not to take risk of throwing good money after bad. This means I needed (haha, I use that word loosely) a new DAC. And after my bad experiences with Topping and SMSL, no more Chi-Fi for me. It’s not worth the risk and hassles. I’ll buy gear from the USA, Japan or Germany for the quality and support.

Back to the Review

Good Stuff

  • Excellent measured performance
  • Great subjective sound quality: clean, detailed, neutral
  • Seamless operation – no software bugs encountered
  • Functionally complete: several digital inputs, useful firmware settings
  • Simple: no controls,  switches or moving parts that can eventually fail
  • Internal power supply – no wall wart
  • Solid build quality
  • Low price

Bad Stuff

  • Single-ended only (no balanced outputs)
  • Line-level only (no headphone amp)
  • Questionable reliability / longevity
  • Poor warranty & support

Overview

First, check out Amir’s detailed measurements on ASR. The SU-6 is a well engineered device and one of the best measuring DACs.

What is the SU-6?

  • DAC supporting PCM and DSD formats
  • Select between 4 inputs
  • Volume control
  • Preamp (line-level single-ended output with volume control)

The SU-6 has 4 inputs, all digital:

  • USB (async/pull)
  • SPDIF (coax and toslink)
  • Bluetooth

Key SU-6 Features

  • Reference quality audio: digital & analog
  • Internal power supply: no wall-wart
  • Digital volume control
    • Perfect channel balance at all levels
    • Preserves high SNR even at low volumes
  • Display shows current sample rate (useful for troubleshooting/confirming)
  • User-selectable digital filters: choose from 7!
    • linear vs. minimum phase
    • sharp vs. slow attenuation
  • User-selectable sample rate sync timing

Case, Knobs & Quality

Overall the SU-6 is a cute little plastic and metal box. And it is little: 5 1/2″ wide, 4″ deep, 1″ high. The connectors feel solid, the display has 5 brightness levels and is evenly lit. Volume is controlled only by remote; there is no knob. To me, this is not a deficiency but a feature since knobs eventually fail, and I wouldn’t use it anyway, as I leave the SU-6 volume set to -12 (-6 dB, to avoid clipping loud content having intersample overs) and control volume with my headphone amp.

In use, the SU-6 doesn’t get warm, but only slightly above room temperature. It has a single capacitive touch button that selects the input. All other functions are controlled by the remote. The SU-6’s only moving part is the power switch in the back.

Volume Control

The SU-6 has a volume control with 100 steps. To assess how the steps interact with volume level, I measured the output level using white noise. The top 70 steps drop by 1/2 dB each (volume setting -70 is -35 dB) and the rest of the steps get progressively bigger until the penultimate step (-98) which is  -71 dB, and the lowest step (-99) mutes entirely.

Settings

Besides volume, the SU-6 has 3 settings controllable from the remote:

  • Digital filters: 1 through 7
  • Sample rate phase lock: 1 through 9
  • Screen brightness: 1 through 5
Digital Filters

The SU-6 has 7 digital filters. TLDR; filter #3 and #6 are the best (flattest frequency & phase response), and filter #4 is the best minimum phase / causal / asymmetric. The filter responses match the descriptions in the manual.

Here’s what I got when I measured them:

Here’s a close-up comparison of filters #3 and #6:

I recommend #6, but filter #3 is a good alternative. Filter 6 has the widest, flattest, smoothest passband response. Its only drawback is that it doesn’t fully attenuate by Nyquist (22,050 Hz). It reaches full attenuation by 24,100 Hz. This leaks HF noise, but it should be benign as all aliases must be > 20 kHz (inaudible). Filter 3 fully attenuates by Nyquist, but in order to do that, it has some passband attenuation (-1.5 dB @ 20 kHz) and ripple (0.1 dB). Both are quite small and inaudible to most people. Some people prefer minimum phase / causal / asymmetric impulse filters. The best such filter in the SU-65 is filter #4.

This goes to show that the CD redbook standard of 44.1-16 is not really sufficient for complete transparency. The sample rate is just a bit too low, making such a narrow transition band that even the best DAC chips (the SU-6 uses the ESS-9038) are not both correct and transparent. And while 16 bits / 93 dB of dynamic range is sufficient for most music, it falls short of the full dynamic range of our hearing. What we really need for complete transparency is at least 48 kHz / 24-bit. This is already a standard with DVD, why not make it the new standard for music too? This would actually simplify things as everything (both movies & music) would use the same format.

Sample Rate Phase Lock

SPDIF is a “push” protocol for digital audio. This means the source delivers audio data and the destination/receiver (the DAC) must synchronize with the source. The problem here is that the source (audio player) & receiver (DAC) each has its own independent clock, and no 2 clocks ever agree exactly. It is not enough for the receiver to buffer and re-clock the received data, because if the clocks don’t exactly match, the buffer will eventually under- or over- flow. So the receiver must compute the average rate at which data arrives, to detect the sample rate in terms of its own clock, use that sample rate even if it isn’t exactly what it’s supposed to be, and constantly re-compute and adjust as it runs.

For example, consider 44.1 kHz sampling. If the clocks are off by 1 part in 1 million, that’s 1 sample every 22.6 seconds. In this case, the DAC must detect this and adjust its clock rate to 44,099.999999 or 44,100.000001.

The SU-6 has a setting to adjust how it does this, a range from 1 to 9. The best sound quality (value 1) is the least permissive of clock differences. This has immeasurable jitter but it will glitch if the clocks don’t match nearly perfectly. Value 9 is the most permissive setting, so it can handle more clock rate variance without glitching. Yet it is more subject to jitter.

How to pick the best setting? Start with #1. If that works, you’re done. If it glitches, increase it step by step until it doesn’t glitch.

If all this sounds theoretical, for those who believe digital audio jitter is a non-issue, consider what I encountered when testing the SU-6. I played frequency sweeps at sample rates 44.1, 48, 88.2, 96, 176.4 and 192 kHz. All of them were smooth and clean except for 176.4, which had a few audible “tics” every time I played it. The sweep looked like this:Distortion measured like this:No question this is audible distortion.

After re-testing and troubleshooting, I couldn’t figure out what the problem was. Then I RTFM and discovered the dP-DLL setting. I increased it from 1 to 2, re-tested and the “tics” were gone. It measured clean, like all the other sample rates:The distortion was gone too, now showing only the distortion from my sound card:

It was cool and satisfying to actually find a problem bench testing this device, then find a way to fix it. Yes, the dP-DLL setting actually does make a difference! And through bench testing I was able to confidently find the best digital filter.