Ayre Acoustics QA-9 USB A/D converter
Then, at the 2011 Rocky Mountain Audio Fest, I was given a dem of what I at first assumed was a QB-9until I saw the bar-graph level meters to either side of the sample-rate display. This was the QA-9, a high-performance analog/digital converter, housed in the same chassis as the QB-9 and intended to allow audiophiles to make rips of their LPs of the highest quality possible.
The QA-9 operates at sample rates up to 192kHz, outputting 24-bit data either by a USB 2.0 connection or by AES/EBU. (The two aren't operational at the same time.) The basic version costs $3950; a Pro version, which includes DSD and Word Clock outputs on transformer-coupled BNC jacks, costs $4750. Given my extensive experience of both domestic and professional A/D converters, which has convinced me that the most critical process in digital recording is the initial analog/digital conversionnothing downstream can put right whatever was done wrong in that conversion (footnote 1)I asked for a production sample of the QA-9.
Via e-mail, I asked Ayre's founder and designer, Charlie Hansen, why a manufacturer exclusively known for its domestic high-end audio products had ventured into a field dominated by pro-audio companies.
"If we build, say, a great preamplifier, then one audiophile will be able to bliss out on the music. On the other hand, if we build a great A/D converter, then literally millions of people could benefit from the improved sound. Several professionals own Ayre equipment, usually for their home systems. Somewhat typical but most notable is Rick Rubin [record producer and founder of American Recordings and Def Jam Records]. He says that Ayre makes the best-sounding digital playback gear, and he has been after us for years to build equally good-sounding recording gear.
"To my knowledge, nobody has ever built an ADC with fully discrete, fully balanced, zero-feedback analog circuitry. In my experience, the single most important factor in getting good sound in digital audio is with the analog circuitry, so that was reason enough to start the project. Of course, we at Ayre are never content to rest on our laurels, so we also incorporated a slew of other innovations."
The heart of the QA-9 is the AT1201, a two-channel A/D converter chip that is new to me, from a company also new to me: Arda Technologies. The chip's datasheet describes it as an "advanced multi-bit sigma-delta" converter that will operate up to a sample rate of 384kHz with an astounding dynamic range of 124dB.
"Arda does consulting work for big contractors. They are audiophiles, and in their spare time designed and built a new A/D chip that clearly outperforms anything from the competition. Specifically, the out-of-band noise is ridiculously low, and this gave us a lot of freedom to 'think different' (as Apple ungrammatically said a couple of decades ago)."
What did Hansen mean by "think different"?
"Ever since DSD was announced, it has received uniformly positive comments on the sound quality. . . . [T]o be honest, we were basically baffled by Sony's original marketing material. We couldn't figure out how the system worked, let alone why it sounded good. But over the years, not only did more and more information leak out, but our understanding of digital audio at Ayre grew rapidly.
"In short, the reason that DSD sounds as good as it does is because there is no filtering done on the record side. The playback side requires a filter (per the Scarlet Book specs), but compared to the brick-wall filters used in typical PCM products, this is a much, much gentler affair. When we developed the QB-9 we spent nearly four months performing listening tests on digital filters. It was clear that both the anti-aliasing (record) and reconstruction (playback) filters had a very strong influence on the sound of digital audio (remember how the 'non-oversampling' DACs were quite popular for a few years?), and so we wanted to really understand what was going on. . . .
"The relatively recent advent of using the personal computer as a way to store and play back music files has changed the game entirely. Now we can buy, store, and play back music at quad-sample rates [176.4 and 192kHz] (or even higher), if so desired. But the problem is that digital audio equipment is still designed by digital audio engineers, traditional engineers, who, if they are using a 192kHz sampling rate, are going to make their equipment with flat frequency response to 96kHz and then just put a brick-wall filter on it. Fortunately, I am not a traditional digital audio engineer! I therefore have the freedom to ask 'Why?,' and I do that a lot!
"So for the QA-9 in Listen mode, we decided that the goal was to make the converter operating at the quad-sample rate to perform more or less like a perfect 30ips analog tape machine. The frequency response is down about 3dB at 50kHz, but it goes down to around 1Hz with no 'head bumps' to worry about. There is zero wow and flutter, and the distortion and noise are about an order of magnitude better (ie, 20dB) than the best analog tape machines.
"Yes, the anti-aliasing filter (in Listen mode) is down only about 20dB at the aliasing point of 96kHz, but is that really a problem? Do any recordings actually have any meaningful amount of musical energy up at 172kHz, where there would be aliasing that would fold down into the audioband?
"Of course not! There aren't many instruments with significant amounts of energy above 90100kHz. Nor are there many microphones with any significant response this high. Or mike preamps. Or mixing boards. Or whatever.
"So the first thing that we did was to use a completely different type of digital filter at the output of the delta-sigma converter. (Every audio ADC chip made for the last 20 years has been a delta-sigma type, as the competing successive-approximation devices died off long ago.) Instead of using the normal low-pass Finite Impulse Response (FIR) filter to turn the output of the delta-sigma DAC into PCM, we use a moving-average filter. This doesn't just allow for improved transient response, but actually perfect transient response. And since we are starting with a 256Fs [11.2896MHz] sample rate, we don't have the problems exhibited by 'non-oversampling' DACs that also have 'perfect' transient response. There is no pre-ringing, no post-ringingno ringing whatsoever. (This is in the Listen mode at both the quad and double sampling rates. It is not possible to use this trick at the single sample rate, where instead we use a more conventional FIR low-pass filter, but of course this is a slow-rolloff design to minimize ringing, and also is a minimum-phase design, so that all of the ringing occurs after the transient, with no unnatural 'pre-echo' before the transient occurs.)
Footnote 1: I haven't forgotten Meridian's implementation of Peter Craven's "apodizing filter," which replaces the original A/D converter's acausal ringing at the Nyquist Frequency with post-impulse ringing at a very slightly lower frequency. But there are many other ways for A/D converters to misbehave.