Technology Preview: NAD M2 Direct Digital Amplifier
Is This The Future of Audio Amplification?
Robert Harley
Note: The full review of the M2, including a detailed description of its sound quality, will appear in the December issue of The Absolute Sound.
The term “digital” is often erroneously applied to amplifiers with Class D (switching) output stages, but in the case of NAD’s new M2 Direct Digital Amplifier that word is appropriate. In fact, the M2 represents a major rethinking of audio-system architecture, directly converting standard-resolution or high-res digital bitstreams into signals that can drive loudspeakers.
Functionally, the M2 is an “integrated amplifier” that replaces a DAC, preamplifier, and power amplifier. The M2 eliminates from a traditional signal path all the electronics of a DAC as well as the active analog gain stages of a preamplifier and power amplifier. It does this by converting the PCM signal from a digital source directly into a pulse-width modulation (PWM) signal that turns the M2’s output transistors on and off. That’s it—no digital filter, no DACs, no multiple stages of analog amplification, no interconnects, no jacks, no analog volume control, no preamp. The conversion from the digital domain to the analog domain occurs as a byproduct of the switching output stage and its analog filter. This is as direct a signal path as one could envision. (See sidebars for the technical details.
NAD’s M2 is a significant departure for the company that made its reputation building simple and affordable electronics. For starters, the M2 costs $5999, a new price level for a NAD “integrated amplifier.” Second, the M2 is NAD’s first amplifier to use a switching output stage. The company had previously rejected the technology in favor of linear amplifiers because switching output stages just didn’t sound good. But the M2’s output stage is significantly different from any other currently offered (see sidebar). Third, NAD believes that the M2’s technology could eventually become the basis for nearly all of its amplification products. In fact, NAD suggested that the M2 was not designed to capitalize on Class D’s functional advantages, but rather to establish a new benchmark of performance in amplification, no matter what the technology.
Let’s look at the M2 Direct Digital Amplifier in operation. The unit looks and functions like one of NAD’s upscale Masters Series integrated amplifiers, with a row of front-panel input-select buttons, a volume control, and a display. The rear panel, however, reveals that the M2 is not a conventional integrated amplifier. Five digital inputs are provided (two RCA, one AES/EBU, two TosLink, plus a TosLink loop) along with one single-ended and one balanced analog input. The digital inputs can accept any sampling frequency from 32kHz to 192kHz. Analog signals fed to the M2’s analog-input jacks are converted to digital. Once you’ve connected an analog or digital source to the M2 (such as a CD transport or music server) and loudspeakers via the output binding posts, the M2 functions just like a traditional integrated amplifier. You select the source from the front panel and control the volume with the large front-panel knob or from the remote control. The front-panel display shows the input sampling frequency and volume setting.
Purists will note that the M2 requires that analog signals, such as a phonostage output, be converted to PCM digital. Similarly, those who enjoy SACD will be loath to convert their SACD player’s analog output to PCM, and then back to analog in the M2.
The M2 offers a number of features not found on a traditional integrated amplifier. Pushing the Menu button allows you to select the sampling frequency of the analog-to-digital converter (for analog input signals) as well as engage an upsampling feature that converts, for example, 44.1kHz to 96kHz. Analog signals are digitized at up to 192kHz/24-bit. You can also attenuate the level of the analog inputs by up to 9dB. A “Speaker Compensation” adjustment is a five-position adjustment that “allows fine tuning of the top octave to match the speaker impedance.” An absolute polarity switch rounds out the menu-accessible features. A rear-panel switch engages NAD’s “Soft Clipping” feature, which limits the output to prevent audible distortion if the amplifier is overdriven. An RS232 port allows external control via a PC or control system such as Crestron or AMX. The full-function remote control selects between sources, adjusts the volume, dims the display, and can also control a NAD CD or DVD player.

The M2 doesn’t seem like a switching amplifier in operation; it is heavier than most Class D amps and although it runs cooler than a traditional Class A/B amplifier of comparable output power, it produces more heat than any other Class D amplifier I’ve had in my home.
Technology: Not Just Another Switching Amplifier
The M2 is different in two important ways from other amplifiers that use a Class D switching output stage. In a conventional switching amplifier, analog input signals are converted to a series of pulses that turn the output transistors fully on or fully off. The signal’s amplitude is contained in the pulse widths. An output filter smoothes the pulses into a continuous waveform. But in the M2, PCM digital signals fed to the amplifier’s input (from a CD transport, music server, or other source) stay in the digital domain and are converted by digital-signal processing (DSP) to the pulse-width modulated signal that drive the output transistors.
This difference might not seem that great at first glance, but consider the signal path of a conventional digital-playback chain driving a switching power amplifier. In your CD player, data read from the disc go through a digital filter and are converted to analog with a DAC; the DAC’s current output is converted to a voltage with a current-to-voltage converter; the signal is low-pass filtered and then amplified/buffered in the CD player’s analog-output stage. This analog output signal travels down interconnects to a preamplifier with its several stages of amplification, volume control, and output buffer. The preamp’s output then travels down another pair of interconnects to the power amplifier, which typically employs an input stage, a driver stage, and the switching output stage. In addition to the D/A conversion, that’s typically six or seven active amplification stages before the signal gets to the power amplifier’s output stage.
To reiterate the contrast with the M2, PCM data are converted by DSP into the pulse-width modulation signal that drives the output transistors. That’s it. There are no analog gain stages between the PCM data and your loudspeakers. The signal stays in the digital domain until the switching output stage, which, by its nature, acts as a digital-to-analog converter in concert with the output filter. The volume is adjusted in DSP.
The second point of departure between the M2 and all other Class D amplifiers is the switching output stage itself. NAD partnered with the U.K. design team of the American semiconductor company Diodes Zetex, who had developed a novel switching-amplifier technology. NAD engineers worked with Diodes Zetex for more than four years to improve upon Zetex’s basic idea before it was ready for the M2. Diodes Zetex calls its amplifier a direct digital feedback amplifier (DDFA). The primary innovation is the use of feedback around the output stage to reduce distortion. Feedback, used in virtually all linear amplifiers, takes part of the output signal, inverts it, and sends it back to the input. The technique lowers distortion. But feedback isn’t practical in switching amplifiers because of the delay involved in sending part of the output signal back to the input. Switching stages operate on extraordinarily precise timing; a glitch of a nanosecond can cause the output stage to lock up. The Zetex innovation is to compare the actual high-level PWM signal (at the transistor outputs) to a low-level reference PWM signal. Any difference between the actual and reference PWM signals represents a voltage error. The actual PWM signal can deviate from the theoretical ideal because of power-supply noise or droop (a drop in voltage), slight changes in the pulse widths, transistor tolerances, or variations in the rise-time of the pulse edges. All these potential sources of errors affect the area under the pulses, which is how the analog amplitude is encoded. This error shows up as a voltage, which is digitized at a conversion rate of 108MHz, processed to compensate for subsequent modulation cycles, and then fed into a noise shaper that adjusts the pulse shape, on a continuous basis, to compensate for errors in the output stage. In addition to decreasing distortion, this technique also lowers the amplifier’s output impedance.
The reference PWM signal must be essentially perfect or else the system will correct “errors” that aren’t present. The pulse widths must be precise to within five picoseconds, a level of performance commensurate with the lowest clock jitter in state-of-the-art digital-to-analog converters. In fact, you can think of the M2 as a DAC with gain and judge its technical performance using the same metrics as those employed in evaluating D/A quality. For example, at -120dB, the M2’s linearity error is less than +/-0.1dB (an amazing spec, by the way), and the unit provides useful resolution down to an astounding –135dB.
The M2’s topology has interesting ramifications for a system’s overall noise performance. In a traditional system of digital source, analog preamplifier, and analog power amplifier, any noise introduced ahead of the power amplifier greatly degrades the system’s signal-to-noise ratio (SNR). For example, if we start with a CD player with a SNR of 115dB, feed its output to a preamplifier with a SNR of 108dB, and then drive a power amplifier whose intrinsic SNR is 115dB (all great specs), the system’s overall SNR is only 84.1dB referenced to 1W (all SNR numbers are un-weighted). Noise at the front of the chain gets amplified by the power amplifier, no matter how quiet that amplifier is. In the M2, the only source of noise is in the DSP and the switching output stage, and the noise level is completely independent of the gain. That is, the SNR doesn’t degrade at low volume. The DSP’s noise is kept low in part because of the 35-bit data path. The M2 has an SNR of 91dB (un-weighted, referenced to 1W) at any signal level. Indeed, I turned the gain all the way up and put my ear next to the tweeter of the highly sensitive Wilson Audio Alexandria X-2 Series 2 loudspeaker (95dB 1W/1m) and heard no noise.
There’s no free lunch, however. Switching amplifiers require a serious output filter (typically a large inductor and a capacitor) to remove high-frequency switching noise from the output, and to smooth the waveform. This filter is conceptually similar to the reconstruction filter in traditional digital-to-analog conversion. Switching amplifiers are also very susceptible to audible degradation if the power supply feeding the output transistors isn’t perfectly clean. That’s because the output transistors either connect the output transistors’ power-supply rail to the loudspeaker (in the “on” state) or disconnect them (in the “off” state). Any noise or ripple on the supply rails is connected directly to the loudspeaker. Switching amplifiers thus require an extremely quiet supply. Nonetheless, many switching amplifiers skimp on the power supply in an effort to keep size, weight, and cost low. The M2 has a more substantial power supply than I’ve seen in any other amplifier with a switching output stage. Three separate supplies are used, one for each audio channel and one for the control circuitry and housekeeping.
Each of the M2’s amplifiers is contained on a roughly 6"-square circuit board and heat-sink assembly that attaches to a mother-board below it. It appears that each channel employs two pairs of output transistors. The rear panel is shielded, presumably to prevent radiated switching noise to get into the signal after it has been filtered. The chassis is segmented into two additional shielded modules, again to protect against switching noise pollution generated by the output stage.
Comments
Come on Robert, you can't leave us hanging like that. Give us a taste of the how it sounds and images. Though it sounds interesting,it seems like a different take on what TACT and Lyngdorf does.
Yes, this sounds essentially the same as what TACT and Lyndgdorf integrated amps do. How is it different?
my intersest is in the First Watt J2,
my intersest is in the First Watt J2,
Robert, were you presenting a question that "could" this be the future? Or do you honestley believe that this will be the future in amplification like music servers/high Rez downloads in the near future? I feel like we are still far from any breakthroughs in amplification. I hope I am wrong.
robert
your kidding me right? - your knowledge is impeccable about all things digital and im impressed about your total understanding - but come on tell us a little about how it sounds - that always is still the bottom line!
The M2 is different from the original TacT Millennium (which I reviewed in 1999 upon its introduction) in several ways. First, the M2's switching output stage is unlike any previous switching output stage (as explained in the technical sidebar). Second, the M2 adjusts the gain in the digital domain rather than varying the voltage of the rails that supply the output transistors. The TacT had certain qualities, but it lacked dynamics. The M2 has no such shortcoming. In fact, it has a sense of explosive power and unlimited dynamic reserves. The M2 also doesn't sound like any other switching amplifier I've heard (although I have not heard many). The M2 is in another league in terms of sound quality when compared with other switching amplifiers. It lacks the characteristic colorations I've heard in previously auditioned switching amplifiers (although I have not heard these colorations in the many excellent show demos by Bel Canto), instead sounding very much like a high-quality linear amplifier. I'll save the full sonic description for the review in the December issue. Thanks to everyone who has responded.
In reality, this IS a DAC, just not in the traditional sense. And because it is a DAC, it is susceptible to all the well-known and unavoidable by-products of 24-bit digital conversion. This is why today's latest DAC technology designers have spent so much time and money on the process itself (apodizing filters with variable slopes and timings to best match program, attention to pre-ringing and group delay, etc..). NAD cannot avoid these well-known issues.
The question is: how did they address the myriad and subtle engineering issues of audio D/A conversion? If they are doing their D/A conversion at the PWM stage itself, where and how are the important low-level bits being managed? How are they filtering? How are they compensating for all the known dynamic complexities of D/A conversion?
To eliminate a high quality dedicated DAC stage (Wolfson, BB, 5-bit flash + discrete, etc.) assumes they are doing it better elsewhere. I don't buy it! Sounds simply like another compromised NAD solution whose price point should be around $3k, not $6k....
Really amazing on how you can prejudge a company and a technology without hearing it. Just who are you a shill for again?
Fair enough. I've owned NAD equipment (3 different pieces) beginning in 1985-ish - a preamp and two power amps. Some NAD gear was a good value. And still is.
But a PWM amp for $6k? Whatever. If you think it's at the right price point, go buy it. But that's not point, and I'm sorry that I wasn't clear.
My point is that NAD is apparently not using a dedicated DAC device, but rather decoding the 24-bit data stream directly at PWM. I simply want to know: how? There are a number of well-known issues with digital conversion that, if not properly handled, will deliver less-than-stellar sound. And at $6k, I would expect nothing but stellar sound. And given that they're not using a dedicated DAC (my assumption based on Robert's review), I would simply like to know HOW they're managing this essential, but complex, conversion stream. (actually, I'm betting they DO have a dedicated DAC device in the signal path..)
p.s... another way to look at this is that, if direct PWM was the most musical method of small-signal digital-to-analog conversion, DAC designers would be converting 24 bit PCM to PWM to line-level analog, rather than SDM.
I might tend to agree with your perspective, but, technology does move forward. Being an Electrical Engineer for the past 20 years, I know that more than most people! It will be an interesting review.
Sharp (of all companies) had a power amp that was very much like this about 7-8 years ago. Bob Jung (engineer extraordinaire) did a review of it in Pro Audio Review and found it to be on par with the Pass Labs amps in his mastering studio. That amp was around $6k, too. Sharp never promoted it, though, and it evaporated.
I'll reserve judgement on this piece until I hear it.
I didn't see a power rating. What's it packing?
2 X 250W Continuous Power at 8 and 4 Ohms
full specs. nadelectronics.com/products/masters-series/M2-Direct-Digital-Amplifier
Sharp made a series of 1-bit products based on that amp.
Most were combo systems however.
Steven Stone
Contributor to The Absolute Sound, EnjoytheMusic.com, Vintage Guitar Magazine, and other fine publications
Converting pcm to pwm is indeed the best way to do it but... for a true linear conversion you need a switching frequency of 2,9 Ghz for 16 bit depth (65536 levels @ 44,1k). For other depths/sample rates go figure... I wonder which is the true switching frequency of the output stage.
"Converting pcm to pwm is indeed the best way to do it ".
Um, why? Can you cite any research on this? Papers? Technical references? Or is this just your opinion?
If PWM is the "best way to do it" -- why do all DAC device makers all use a SDM variant? Why not convert PCM => PWM => linear analog?
It could be argued that the "best way" to get analog program in and out of the digital domain is high-bit DSD.
One problem with SDM and, in particular, the DSD in the output stage is the high frequency: most practical high-power gates have finite switch times, consequently, the higher the switch frequency - the less efficient the gate becomes.
On the other hand, PWM modulators used in audio amplifiers operate on frequencies roughly one tenths of the one of the DSD and hence the switches from rail to rail occur much less frequently. As a result the time the gates spend in transition, and hence convert the electrical power to heat, is reduced considerably.
Of course, this does not come for free and a designing a linear PWM modulator by itself is not at all trivial. Luckily, the problem was solved more than 10 years ago and the only thing that was missing was an essentially analog feedback topology that would allow the correction for the gate switch timing errors, power ripple, etc. yet would not completely undermine the whole idea of the “fully digital” power stage.
The folks from the company bought by NAD seem to have found an elegant solution, which, in my opinion, is quite obvious and it is really odd that no one came up with it sooner.
At a practical level, surely NAD have made a mistake by not including an ethernet connection. If one could stream music to it like the Linn DS range then perhap's you'd have a superb amp and dac for half the price of the Linn Klimax.
yes definitely the amplification is the future of not only the audio but also the cellular amplification..as this is the need to accomplish the certain criteria as the people like the better audio as well as better cell signals. so we need better audio and cellular amplifier.
thanks .
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