PS Audio PerfectWave DirectStream MK2 D/A processor Measurements

Sidebar 3: Measurements

I measured the DirectStream MK2 first using my Audio Precision SYS2722 system, then with the magazine's higher-performance APx500. The review sample was fitted with the v2.3.5 FPGA firmware, but after I experienced some anomalous measured results, I installed the v2.3.3 firmware package, which was available as a download on the DirectStream MK2's information webpage. I then installed the v2.3.6 firmware when it was available and repeated the testing.

The DirectStream MK2's optical input locked to sample rates up to 96kHz, the coaxial and AES3 inputs locked to datastreams with all sample rates up to 192kHz. Both Roon and Apple's USB Prober app identified the USB-connected D/A processor as "PS Audio Extended Resolution USB" from "PS Audio," with the serial number string "Version 3.02 11/30/2022." USB Prober revealed that the USB port's "Primary Play Interface" operated in the optimal isochronous asynchronous mode. Apple's AudioMIDI utility also identified the PS Audio as "Primary Play Interface" and indicated that the DirectStream accepted 16-, 24-, and 32-bit integer data via USB sampled at all rates from 44.1kHz to 705.6kHz.

With the volume control set to its maximum, a 1kHz digital signal at 0dBFS resulted in an output level of 3.72V from the balanced outputs, which is 2.3dB higher than that of the original DirectStream. The maximum output level was 3.7V from the single-ended outputs when these were set to "Balanced," which was how the review sample had been set up. (In this mode, the unbalanced output is taken from the hot and cold transformer connections.) Using the Menu to set the single-ended outputs to "RCA Unbalanced" reduced their maximum output by 6dB, as expected, to 1.86V. The volume control operated in accurate 0.5dB steps, and the DirectStream MK2 preserved absolute polarity, ie, was noninverting, from both output types with the Phase switch set to "Norm." The balanced output impedance is specified as <200 ohms; I measured a still-low 400 ohms at 20Hz and 1kHz, dropping to 369 ohms at 20kHz. The unbalanced output impedance was lower, at 175.5 ohms, in the bass and midrange and 166 ohms at the top of the audioband. (The specified unbalanced impedance is <100 ohms.)

Fig.1 PS Audio DirectStream MK2, impulse response (one sample at 0dBFS, 44.1kHz sampling, 4ms time window).

Fig.2 PS Audio DirectStream MK2, wideband spectrum of white noise at –4dBFS (left channel red, right magenta) and 19.1kHz tone at 0dBFS (left blue, right cyan), with data sampled at 44.1kHz (20dB/vertical div.).

The PS Audio's impulse response with 44.1kHz data (fig.1) indicates that the reconstruction filter is a conventional linear-phase type, with time-symmetrical ringing on either side of the single sample at 0dBFS. As DSD data can be converted to analog with a simple low-pass filter, this impulse response is due to the resampling filter earlier in the processing chain. (In the video presentation mentioned in footnote 2 in the review text, Ted Smith says that the FPGA he uses allows him to implement a long digital filter with a large number of coefficients.) With 44.1kHz-sampled white noise (fig.2, red and magenta traces), the DirectStream MK2's response rolled off sharply above 20kHz, reaching full stopband suppression at half the sample rate. The level of the ultrasonic noisefloor starts to rise above 40kHz, which will be due to the PS Audio converting the PCM data to DSD. There are no aliased images of a full-scale tone at 19.1kHz (blue and cyan traces), though some distortion harmonics are visible, with the third lying at –80dB (0.01%).

Fig.3 PS Audio DirectStream MK2, frequency response at –12dBFS into 100k ohms with data sampled at: 44.1kHz (left channel green, right gray), 96kHz (left cyan, right magenta), and 192kHz (left blue, right red) (1dB/vertical div.).

When I examined the PS Audio's digital frequency response with AES3 data at 44.1, 96, and 192kHz, the response at all three rates dropped off sharply just below half of each sample rate (fig.3). The balanced output is down by just 0.5dB at 20kHz; the unbalanced outputs behaved identically, unless set to "Balanced," when the output was –1.9dB at 20kHz. Channel separation (not shown) ranged from 70dB at the top of the audioband to 80dB at low frequencies.

When I examined the level of the PS Audio's noisefloor, I ran into anomalous results. The upsampling to DSD results in a relatively high level of ultrasonic noise—using the TosLink optical input, I measured 210mV of noise with a center frequency of 380kHz present in the balanced outputs. By contrast, the original DirectStream DAC fitted with the "Windom" FPGA firmware had just 11mV of ultrasonic noise with a center frequency of 283kHz in its balanced outputs (footnote 1). Both Audio Precision analyzers can apply a "brickwall" low-pass filter with a cutoff frequency of 20kHz or 40kHz, but with this filter, I wasn't able to get consistent results with the MK2 until I updated the FPGA firmware.

Fig.4 PS Audio DirectStream MK2, balanced output, spectrum with noise and spuriae of dithered 1kHz tone at 0dBFS with 24-bit data (left blue, right red) (20dB/vertical div.).

Fig.4, therefore, shows the DirectStream's low-frequency noisefloor after the change in FPGA firmware as it drove a full-scale 1kHz tone from the balanced outputs, with the volume control set to its maximum of "100." The level of the random noise is about 30dB higher than I was expecting, and if I lowered the volume by 6dB the noisefloor rose by the same 6dB (footnote 2). This graph was taken with the SYS2722; repeating the spectral analysis with the APx500 gave a similar result, even with the brickwall filter operating. The level of the ultrasonic noise wasn't affected when I used the Menu command to lift the output ground or that of the XLR shells.

Fig.5 PS Audio DirectStream MK2, left channel, 1kHz output level vs 24-bit data level in dBFS (blue, 20dB/vertical div.); linearity error (red, 2dB/small vertical div.).

Fig.6 PS Audio DirectStream MK2, spectrum with noise and spuriae of dithered 1kHz tone at –90dBFS with 24-bit data (left blue, right red) (20dB/vertical div.).

The presence of this noise affected my measurement of the PS Audio's linearity and resulted in a rising error below –80dBFS (fig.5). However, when I looked at the spectrum of the DirectStream's output while it decoded dithered 24-bit data representing a 1kHz tone at –90dBFS, the tone was reproduced at the correct level (fig.6). The apparent positive linearity error at low levels must therefore be due to the noise.

Fig.7 PS Audio DirectStream MK2, waveform of undithered 16-bit, 1kHz sinewave at –90.31dBFS (left channel blue, right red).

With undithered 16-bit data representing a tone at exactly –90.31dBFS (fig.7), the three DC voltage levels described by the data were obscured by high-frequency noise.

Fig.8 PS Audio DirectStream MK2, balanced output, 24-bit data, spectrum of 50Hz sinewave, 10Hz–1kHz, at 0dBFS into 100k ohms (left channel blue, right red; linear frequency scale).

Fig.9 PS Audio DirectStream MK2, balanced output, 24-bit data, HF intermodulation spectrum, DC–30kHz, 19+20kHz at 0dBFS peak, sampled at 44.1kHz.

As suggested by fig.2, the third harmonic was predominant but slightly higher in level at low frequencies, at –70dB or 0.03% (fig.8), than it was in the midrange and treble. The intermodulation products with an equal mix of 19 and 20kHz tones with a peak level of 0dBFS mostly lay around –90dB (0.003%, fig.9), though a series of products spaced at 1kHz intervals can be seen.

Fig.10 PS Audio DirectStream MK2, high-resolution jitter spectrum of analog output signal, 11.025kHz at –6dBFS, sampled at 44.1kHz with LSB toggled at 229Hz: 16-bit AES3 data (left channel blue, right red). Center frequency of trace, 11.025kHz; frequency range, ±3.5kHz.

The high level of random noise meant that I couldn't meaningfully examine the PS Audio's rejection of word-clock jitter. For example, fig.10 was taken with 16-bit AES3 J-Test data and the Audio Precision brickwall filter set to a 40kHz bandpass. Nevertheless, all the odd-order harmonics of the LSB-level, low-frequency squarewave lay beneath the noisefloor. Peculiarly, the level of random noise in the right channel (red trace) is lower than in the left channel (blue). The levels of the noise became equal if I bypassed the low-pass filter but were about 15dB higher than they were in fig.10.

In most ways—the behavior of the reconstruction filter, the frequency response and channel separation, the sufficiently low levels of distortion—the PS Audio's measured performance is respectable. But in others—the apparent linearity error and the rejection of jitter—the results were dominated by the presence of noise in the DirectStream MK2's output, mostly ultrasonic but in-band, too. This is concerning because amplifiers with a wide ultrasonic bandwidth but poor compensation will suffer more in-band modulation noise, which may impact subjective performance (footnote 3).—John Atkinson

Footnote 1: The original DirectStream's measurements can be found here, here, and here.

Footnote 2: In other words, the absolute level of the noise was not affected by the volume-control setting.

Footnote 3: The main concern, specifically, is intermodulation of ultrasonic noise or signal-correlated products, into the audible range, which could be the reason for the in-band noise in the DirectStream MK2. Signal-related IM products, if they occur at audible levels, are likely to be more objectionable than noise.—Jim Austin

PS Audio
4865 Sterling Drive
CO 80301

hollowman's picture

In one of the PS Audio YT videos, Paul M shows the lobby showcases at PS Audio HQ in Colorado. In them are all the products PSA has developed over their history.
For DACs, there have been several UltraLink models. Here's a Stereophile review from 1995:
It would be curious to re-review a fully-working classic DAC -- and compare/contrast w/the latest DAC.

John Atkinson's picture
hollowman wrote:
It would be curious to re-review a fully-working classic DAC -- and compare/contrast w/the latest DAC.

In 2019 Herb Reichert compared the original PS Audio DirectStream DAC with the 1989 PS Audio Digital Link. See

John Atkinson
Technical Editor, Stereophile

hollowman's picture

JA: thx, for the link to the Link!
And I did see that you also measured that old dac in 2019 with modern metrological instruments, like the SYS2722.

The Digital Link used the then-new Burr-Brown 18-Bit DAC chip, the PCM 61P, in a dual set.
In the Link, the entire d/a chip set seems to be:
2 x PCM61P - YM3434 - YM3623B

Image here:
Some possible issues with Herb Reichert's 2019 comparison might be the age/condition of those eletro caps (are they orig. in his re-review unit ?). Also, the lack of I2S direct input may have compromised sonics between the two units.
As a far as JA's note about Most Significant Bit (MSB) adjustment ... first thru third gen Philips dacs (up to Bitstream) dealt with that issue "expensively" via DEM (dynamic element matching) and external ceramic coupling caps (e.g., the tda1541, on the 3x2 bit active divider pins). Later multi-bit dac chips addressed the MSB issue internally (Philips Continuous Calibration) , as AD did with their AD1862, and trimmer pins, as does the PCM61P . Modern R2R discrete dacs, deal with this via dsp control.

Nota Bene:
Re-measuring (and re-reviewing) older, well-kept gear is a very good idea! It keeps the new-equip manufs in-check; and the 2nd-hand communities (AudioKarma, Audiogon), objectivists (Hydrogenaudio ) and DIY communities all interested and engaged in what mainstream hifi media are ... ahem ... "agenda-ing."


MhtLion's picture

Subjective listening and the musical enjoyment is another thing. But, how I interpret the provided measurement here of PS Audio PerfectWave DirectStream MK2 D/A is that this DAC is not worth its asking price. Personally, these measurement speakes that PS Audio does not have the industry leading engineering pedigree when it comes to a DAC.

I'm not saying this is a bad sound DAC. Not at all because I haven't heard it. But, in order for a company to say 'we know a thing or two about a DAC' it first need to produce a good measurement or at least very good at making elaborated BS claims why they product intentionally sucks at the measurement, which apparently some people buy. Have you tried a popular DACs under $900 with a $100k system? Schiit, Topping, SMSL - They are sounds good playing with systems 10X of their retail price. To say a DAC sounds good - doesn't mean much. I don't remember any bad sounding DAC above $500 in last a couple years. To get a merely sounding good DAC, you don't have to spend $8k. For $8k - it needs to be special. It needs to so good that once taken out of system you miss it, cannot stand without it.

Glotz's picture

But the various online audio communities tell everyone they do. Largely, because they don't have experience listening to the gear, and measurements 'tell' them they don't need to listen.

Yet, when everyone actually listens exhaustively to the SMSL and the Topping DACs compared to $10k plus units, they suffer in image size, depth of field perspective and focus.

Any DAC can be placed in a $100k system and will sound pretty great. It's the DAC at $10k and up that need to justify their position in those systems... and do. They prove it in the listening. There would not be a market for those expensive DACs if not.

By price and measurements, though excellent, you still dismiss the PW DAC.

For a DAC to be 'special', you haven't noted any parameters for such, other than price and measurements.

hollowman's picture

I’ve gone thru Ted Smith’s videos and posts about how DSD “is” the “analog” signal and all one really needs to do is LP filter. That’s a very simple interpretation; please correct me as needed!
I was going thru the HiFiEngine’s schematics and serv. manuals and ran across the Arcam Black Box “Delta” series of DACs.

And comparing to the master digital chip list on:

That dutchaudioclassics list has the wrong chipset for the Black Box 500 DAC. Going thru various Black Box schematics, Arcam did what I believe Ted suggests--way back in 1993!!! The TDA1307 is a rare and unique DF, made by Philips, and used various Philips / Marantz products --- including the very high end Marantz SACD unit from 2000.

The 1307 got little attention because the PMD100 chip with HDCD was the hot, attn-grabbing rock star in the mid/late 1990s. The TDA1307 interpolator converts I2S (PCM) to Bitstream (“DSD”). And then Arcam follows that with their own, custom “Bitstream digital to analog converter (DAC)” which may be what Ted had in mind. Not sure. The schematics for the Black Box 500 are readily avail at the usual places. Have a look!