Articles, Reviews and Measurements of Audio Products

Here is my coverage of Day 2 of 2017 Rocky Mountain Audio Fest. As this was a full day, there was a lot more to cover. Hope you find it enjoyable and possibly informative.

Hello you all. So the day 1 at the Rocky Mountain Audio Fest (RMAF) came to a close. It was a calm and "orderly" day with little to gripe about . Being part of the press, I did not have to wait in this line where last year they could not find my badge.



The line had died a few hours later.

The rooms had light to medium attendees. But when I asked a few companies what traffic they were seeing, they all said it was quite high and they were very satisfied with it seeing how it is not the weekend.

The weather outside is perfect with leaves starting to turn and super comfortable temps. But what the heck is going on here???


Monday is the day I fly out! How do we go from 70s to 30s with snow and then back to near 70s???? It does not compute!

Had great number of conversations with industry folks. Unfortunately i can't share much of it with you all.

Overall, nice, relaxing and...

The Henry Engineering Matchbox II was loaned to me by a member for measurements and hardware teardown/review. I am not sure how well known this DAC is. It is targeted toward television broadcast world. It unusually has both ADC and DAC plus professional AES/EBU output (balanced version of S/PDIF). In that regard, it appropriately only has balanced inputs and outputs.


Retail price is fairly high for consumer market but appropriate for professional use at $479.

Specs are rather modest, stopping at just 48 Khz sampling and bit depth of 16 bits. Again, this is fine for video applications where 48 Khz sampling is standard and 16 bits sufficient.

Measurements
As always, my go-to measurement is 24-bit, 48 Khz J-test signal. I had not noticed that this device is limited to 16 bits when I did the testing so the 24-bit depth of the signal is lost on it. Indeed, that shows up in the measurements:



The...

I was recently asked about merits of S/PDIF versus USB for audio DACs. It has been said that S/PDIF was designed for audio whereas USB is a computer interface. And that makes USB noisier and less desirable interface.

I think most of you know my opinion on this. But just in case, I believe USB to be a superior and more "correct" interface for audio. Problem with S/PDIF is that it makes the source the "master," forcing the DAC to chase its timing. This means that if the source S/PDIF signal is not very clean, it can impact target DAC performance. Fortunately over the years S/PDIF interface has been perfected a lot and even in low cost implementation it can be excellent.

Still, there is no reason to have this antiquated architecture. Using asynchronous mode USB, the DAC can set the cadence using a high-performance clock and force the source, in this case a computer or streamer, to follow it.

Yes, there is some risk of noise here as USB is a much more complicated interface...

This is a review of the Musical Fidelity V-DAC II including measurements. Member Ron Party was kind enough to l0an this to me. This is an older DAC, dating back to 2012 or so and retailed for $379. It comes with an external power supply which looks to be a linear one.

As usual, my first measurement is J-Test signal at 48 Khz sampling and 24 bits. Here is the outcome as compared to iFi iDSD which is a much newer DAC at similar price point:



As we see there, the noise floor of Musical Fidelity V-DAC II is lower than iFi iDSD even though its output is higher (i.e. has a better signal to noise ratio). Just eyeballing it, it seems to have 5 to 6 db advantage over iFi.

Given the linear power supply, its output is free of mains related harmonics. So overall, a very nice showing here.

I have also shown the performance of V-DAC using both its USB input and S/PDIF. S/PDIF was generated from my USB port using an Audiophilleo USB to S/PDIF converters. As...

Note: this article was originally written by our DAC/technical expert, @DonH56.

The previous thread on DAC fundamentals provided an overview of conventional (Nyquist) DACs. This thread will introduce oversampling and the delta-sigma architecture that dominates the DACs used in consumer audio gear today.

A few definitions:

Nyquist = fs/2 = 1/2 the sampling frequency. This is the highest frequency that a sampled system can correctly capture and reproduce. Any higher, and the frequency information is lost. Note that Nyquist applies to the highest frequency in the signal, so an audio system can reproduce a 20 kHz sine wave ( a single tone) but not a 20 kHz square wave (which has many higher harmonics). A system sampling at frequency fs, e.g. 40,000 cycles per second (40 kHz), can acquire up to (but not including) 20 kHz signals...

Author: our resident expert, DonH50

The purpose of this thread is to provide a quick introduction to digital-to-analog converters (DACs), the magical things that turn digital bits into analog sound. Previous threads have discussed sampling theory, aliasing, and jitter. Now we’ll get down to the hardware and take a look at some basic DAC architectures.

The two criteria most often used to describe a DAC are its resolution (number of bits) and sampling rate (in samples per second, S/s, or perhaps thousands of them, kS/s). If we think about producing an analog output both of these are important. The resolution determines the dynamic range (in dB) of our DAC, and sampling speed determines how high a signal can be output. As discussed in those earlier threads, resolution sets an upper limit on signal-to-noise ratio (SNR) and spurious-free dynamic range (SFDR, the difference between the signal and highest spur). For an ideal (perfect) DAC, we can find:

SNR = 6.021N + 1.76 dB; and,
SFDR...

Author: our resident expert, DonH50

We left Digital Audio Jitter Fundamentals talking about digital signals. However, error correction and design margins mean jitter on the digital bit stream is rarely an issue for the bit rates used for A/V systems. (At 10 Gb/s and above, it is a bigger issue.) When jitter is brought up as an issue in the audio world, we are talking about jitter on the sampling clock. This can happen for all the reasons mentioned before, but once that clock is used to drive your DAC, the jitter goes right to your ears (OK, there are a few steps along the way, but you get the idea).

Clock recovery is a complicated subject beyond the scope of this thread. Let’s just say getting a very clean, low-jitter clock takes some effort. As a result, jitter can run pretty high (several ns or more) in many audio systems. Make it an A/V system with...

Author: our resident expert, DonH50

Units, Symbols, and Terms, Oh My!
I had intended this to be a short overview of basic terms. Didn't work out that way, sorry. At least it may help in defining some basic terms for people who have not seen them, or have seen them and never knew what they meant. More likely it will simply bore us all to tears. Oh, well! - Don

Before starting a discussion on sampling and such, I thought I should define some of the units, symbols, and terms most of us use but perhaps not everybody understands. I am paraphrasing many of these terms in an attempt to be clear and concise for the non-EE’s among us, not to be completely rigorous, so if you know better please bear with me…

I = current in Amperes (A). This tells how many electrons per second are flowing through a wire (let’s leave RF fields out of it for now).

R = resistance in ohms (Ω). This is the resistance to current flow and is...

Author, our resident expert DonH50

Jitter 101
Jitter is yet another one of those things not terribly hard to understand but with lots and lots of nuances and seemingly hidden details. To begin, let’s define some terms, starting with aperture time.



Figure 1 shows a signal we want to sample. It moves fastest (has the highest slew rate) at the center crossing. Zooming in, the time it takes the signal to change by 1 least-significant bit (lsb) in amplitude is the aperture time (tap). If we sample anywhere within this time window, the output will be the same code (assuming the amplitude starts and ends on a threshold). If we fall outside the window, the next code lower or higher will be output. For a sine wave of amplitude A and frequency f, the maximum slew rate is 2*pi*f*A (the magnitude of the first derivative with respect to time). For an N-bit ADC or DAC and that same sine wave input, the aperture time is 1/[(2^N)*pi*f] (the...