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.
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.
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:
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...
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.
Oversampling refers to how much "extra" bandwidth, or sampling rate, we have relative to the Nyquist rate. For the...
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:
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...
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...
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...
Author: our resident expert, DonH50 (above title mine )
Building a Square Wave
Since I used a square wave as an example in another thread, I realized that what is common knowledge for hairy-knuckled engineers such as I and high-brow scientists like some of the other folk here, may still be mysterious to many audiophiles. I thought it might be worth recreating a simple set of plots that shows how we can make a square wave from a bunch of simple sine waves, i.e. single frequency tones.
For starters, we must realize that any signal can be represented by an infinite sum of single tones of the right amplitude and phase. This is a fundamental principle upon which all signal processing is based. Problems arise, like in many areas of life, when reality hits the theory... In this case, it's impossible for a real system to have infinite frequency response, sampled or not, and of course getting all those tones' amplitudes and phases just right when we add them up is...
Here's an attempt to explain aliasing -- the frequency folding that happens whenever you sample a signal. As was discussed in the Sampling 101 thread, whenever you sample a signal at a rate of X samples/sec (X S/s), the highest output signal is < X/2, the Nyquist rate. That is, when sampling at fs, any frequency equal to or greater than fs/2 will be aliased to fall with the frequency band from 0 to < fs/2.
For CD-rate sampling at 44.1 kS/s, we can convert a signal no higher than 22.05 kHz, or aliasing will occur. If the ADC has sufficient bandwidth, it can capture a signal higher than that, but it will be folded back (aliased) into that 0 - 22.05 kHz region. If the ADC is perfect, the amplitude and phase will be unchanged, but the frequency will be reflected about the Nyquist frequency, 22.05 kHz. A picture may help:
This picture shows a frequency (x) axis with the Nyquist frequency (fs/2), sampling...
Sampling is a very complex problem but the basics are not all that difficult to understand. We’re simply taking samples of things, audio signals in this case, at a particular rate and resolution (terms to be defined in context shortly). Let’s start with the Shannon Sampling Theory, the basis for digital audio:
“If a function of time f(t) contains no frequencies higher than W Hertz, it is completely determined by giving the value of the function at a series of points spaced 1/2W seconds apart.”
Claude Shannon was researching information theory and the most efficient ways to transmit and recover information. Harry Nyquist, a controls expert, refined the theorem and gave us the well-known Nyquist frequency limit: the maximum bandwidth that can be captured by sampling at frequency fs is fbw < fs/2.
A few comments on the sampling theorem:
It applies to an infinite series of samples of infinite precision (i.e. “analog” samples, or...
Wavelength vs. Frequency
Nothing special in this one, just a simple plot. When discussing distances, whether related to room interaction, comb filter effects, room treatment or whatever, there is often the assumption that bass frequencies are "everywhere" while higher frequencies can be "directed" or are "more directional". The reason for this argument has to do with wavelength -- the length of the total sonic wave at a given frequency. Like waves in the ocean, there is a certain distance from peak to peak of audio signals, and that is their wavelength. It is related to frequency; higher frequencies have shorter wavelengths. The actual equation is w = vp / f where w is the wavelength (e.g. feet), vp the propagation velocity (about 1130 ft/s for sound in dry air at sea level), and f the frequency (Hz). Room modes or comb filter effects happen when sound waves bounce off walls or interact with other surfaces, or even arrive...
This is a normal looking USB cable going in and out of a metal box that is about 2 inches long and 1.5 inch wide and 1 inch deep.
The one I have is gray and has no markings whatsoever. But otherwise looks the same, sans the extra length of the one I have. I estimate it to be around 2 meters/6 feet which their web site says retails for 360 euros. ex-VAT. That is about $428 at today's exchange rate. So not cheap at all.
The setup is as with the other thread with Sonore microRendu as the source (networked) player and the DAC, Schiit Modi 2. Here is a comparison of against generic long USB cable...
Aftermarket USB cables seem to be all the rage with people paying as much as $15,000 for a short run of it!!! Lots of subjective performance reports are out there but none back with any objective measurements that demonstrates any difference. On the other side of the fence many believe there isn't or can't be any difference in USB cables ("they are all digital"). I thought it would be good to add some data to the conversation.
For DAC, I used our perennial favorite, the Schiit Modi 2 DAC. Of course I say that in jest as this is by far the worst DAC I have tested. It seems to be highly sensitive to power and USB conditions. I figured if there is one...
This is a review measurement of an Ethernet isolator cable (passive) which is sold to medical industry. Ethernet is normally an isolated interface with transformers at both end. Addition of this is likely for high voltage surges and such.
The unit retails for $204 in US and about 153 euros.
Searching on CA forum, seems like folks advocate its use to get rid of leakage noise. So I used the Sonore microRendu networked audio adapter which seems to be pretty sensitive to AC mains leakage. To make sure it didn't contribute any, I used my lap supply to power the microRendu.
The Ethernet connection is from my local lab switch which in turn connects to another larger switch in basement equipment closet. The cable from switch is about 6 feet. The isolator was placed at the end of the cable and terminating into...
Last year I reviewed the original version of networked audio streamer, Sonore microRendu. Recently someone offered to loan me the newest version 1.4 to test and measure. So here are the results.
The Sonore microRendu is a super light, tiny aluminum box, requiring external power to operate. The device is too light to support the weight of the cables attached to it and runs super hot. I don't recall the original version running this warm. I can barely keep my hand on it for a few seconds.
The microRendu comes with support for Roon which is the way I tested it. For external power, I tested it with iFi iPower power supply ($50), UpTone LPS-1 and my Lab Power supply.
Roon recognized the device after a minute of so of being powered on and reliability was good.
So as a way to network a DAC and remotely using it from your computer or NAS device, it works fine.
Retail price is $650 which is quite high. You can easily buy a computer for the same price...
Here, I am tearing down the box and showing the guts. Hope you enjoy .
The unit is rather small but pretty heft. The shroud is a heavy gauge anodized aluminum which gives me no doubt about its ability to dissipate the necessary heat. After taking off the very tight torx screws, we are greated to this beautiful sight:
As you see, it is a two-board configuration. The small board on top is the switchmode DC to DC converter (more on this later). The base board holds the microprocessor (most likely ARM based) and all the necessary peripherals.
I am a sucker for white solder mask and hence my comment about beauty. It doesn't do anything electrically but after decades of seeing green ones, it is an emotional relief.