Holo Audio May NOS DAC with oversampling on demand
Feb 7, 2020 16:44 · 2643 words · 13 minute read
It’s a DAC that can be operated without or with oversampling and does not only do PCM but also DSD in the fully discrete R2R type digital to analog converter. It looks like we can have our cake and eat it too…. Holo Audio May is a remarkable DAC. Not only because it is a non oversampling DAC using ladder converters but also since it can be switched to oversampling and has an impressive power supply in a separate housing. Something that’s rare for products under € 5,000. The May is their flagship model and it comes in three flavours: Level 1, Level 2 and KTE. Level 2 uses higher quality components while the KTE version goes even further and has, for instance, 4N silver transformer windings. I had the Level 2 version visiting me.
01:04 - As said the May comes in two boxes, one holding the power supply while the other contains the DAC itself. A compact metal remote control is included. The power supply is connected to both the mains outlet and over a proprietary cable to the DAC. The analog outputs of the DAC are connected to an amplifier over either RCA or XLR cables. When you want to play music from your PC or laptop, just connect a USB cable between the computer and the May. If your source is CD, connect the cd-player to the DAC over either I²S, using an HDMI cable, AES/EBU using a 110Ω XLR cable, SPDIF using either a 75Ω RCA cable or 75Ω BNC cable or TOSlink using an optical cable.
In general that is also the priority for quality, provided 01:58 - the source has the identical output: USB, I²S, AES/EBU, SPDIF and TOSlink. But the quality of the respective outputs on the sending device might change that. The remote lets you select the appropriate input, mute the output, dim the display and set oversampling modes, on which later more. The volume buttons on the remote do not work on the May. It is not clear if this will change with a firmware update.
If you want to use a streamer or network 02:29 - bridge you need to connect it to your network on one side and to the DAC on the other side. Here again the preference of interface is the same. When using a streamer or network bridge, it is often controlled over a tablet or smartphone. When you want to use a network bridge, you need player software on the computer that is able to send music to the network bridge. Some popular brands are Logitech Media Server, Audirvana, JRiver and Roon. DLNA servers also do a good job, the best I know is MinimServer. The two boxes are identical, apart from the front, of course. They are built from machined aluminium with brass coloured side panels that, together with the brass buttons on the front give them a luxury appearance. Their width is 430 mm, the depth 300 mm and the hight of each box is 67 mm including the damping brass feet. The weight of the total set is a mere 15 kilos.
The front of the power supply holds no knobs or lights. The power is switched on and off at the rear of the power supply. There also is a power button on the front of the DAC. As far as I can see, this really switches of the DAC and is no standby function. Central on the DAC the display that in normal operation shows the sampling rate, the input and the sampling mode selected. On the right four buttons for mute, display dim, oversampling and input select. Looking at the rear we see the IEC mains inlet with next to it the aforementioned power switch and further to the right a Lemo-like multipole connector that outputs the low voltage DC power. This is to be connected over the supplied bespoke 90 cm long cable to the DC input on the DAC. There are two SPDIF inputs, one on RCA and one on BNC. The latter being the better option if the source also offers BNC and you use a 75Ω BNC cable.
The two 04:37 - other serial inputs are AES/EBU on XLR connector and optical on TOSlink. The USB B connector accepts USB Audio Class 2 that now is supported by all current computer and mobile operating systems. It is also the standard used on modern streamers and network bridges, provides that they have USB outputs. The last two inputs are I²S inputs that use HDMI connectors. Be aware though that you can’t connect these to for instance a TV or blu-ray player. You do need a source that offers I²S as output on HDMI. And then even you have to make sure it uses the same de facto standard as defined by PS-Audio for there are other variants too. But, at least in theory, it is the best digital interface given the separate data and clock signals. An HDMI cable is used since it is designed for extremely high bandwidth and is readily available. Again, don’t use these inputs to connect to video equipment.
The 05:39 - analog outputs are available balanced on XLR cable and single ended on RCA. This time we have two insides. Let’s start with the power supply. The most eye catching items are two 100VA O-type transformers. These use a round core in contrast to metal bands rolled to a core as in toroidal transformers. The sharp edges of that core pack not only increases the risk of broken winding during production, they also cause a less constant magnetic stray field. The O-type transformer was new to me but transformer specialist Menno van der Veen helped me out here.
There are also two 15VA toroidal transformers that remarkably 06:26 - are labeled ‘ringkerntrafo’, the Dutch for toroidal transformer. Also remarkable is the 220 volts print on the voltage selector. This should have been 230 volts. These switches connect two primary windings of a transformer in parallel or in series. With the transformers used here that makes them suitable for either 115 or 230 volts. So there is no problem here, especially since these switches are not to be operated by the user.
It is, by the way, 06:56 - the first time I see a total of 230 VA over four transformers to power a single two channel DAC. Time to go to the DAC itself. It has three compartments, the middle holding a lot of filtering, the USB interface board with XMOS USB receiver. Below that the electronics for the other digital inputs, including an upsampling chip by AKM. And then in the middle the system processing with two Altera MAX V FPGA’s These are processors that are programmed by Holo Audio themselves. The two outside compartments hold the actual ladder converters and analog electronics.
The ladder converters are on separate boards, one for each channel. They are controlled by Altera MAX II FPGA’s that control these banks of switches that switch these resistors to the summing bus. Before I explain that to you, let me point you to these large 1µF capacitors in the output circuits. A ladder converter is a circuit that holds a series of resistors. A 16 bit converter holds - in theory - 16 resistors, each one feeding half the voltage of the previous resistor to the summing line.
If all switches are on, this - according to the Red Book standard 08:19 - - results in 2 volts output. By the way, you can’t measure audio signals with a multimeter, but I liked the graphics this way. If all switches are off, there is 0 volts output. If the digital output signal is a 1 - we call that the most significant bit - followed by 15 zero’s the output will be half the total output voltage, 1 volt. If the digital signal is 01 followed by 14 zero’s, the second significant bit activates the second switch that is designed to produce half the voltage of the MSB switch, so.5 volts. The third significant bit brings.25 volts and so on. The MSB stands for 1 volt while the sum of all other switches stand for the other 1 volt.
By combining a number of switches any voltage 09:09 - between 0 volt and 2 volts can be output in 65,536 steps. If we go to 20 bit resolution, the LSB should add 1.91 µV. Thermal noise and the precision of resistors makes it virtually impossible to stretch the resolution to more than 20 bits. DAC’s that I have measured over the years never achieved a real world resolution of 21 bits and that was up till now and including oversampling DAC’s. The only reason we have 24 bit DAC’s is because processors like to think in groups of 8 bits.
But the lower four bits have absolutely no 09:46 - relevance in distribution media. Not that this is a problem. If a DAC is well built, a 20 bit resolution sounds extremely good. Achieving that resolution with a ladder converter is not straight forward. The tolerances of the resistors and other components will always cause non linearities at low levels. But for every problem there is a solution. In this case even more than one. The most popular way is to stack a number of ladder converters on top of each other.
This way the imperfections of one ladder converter is middled out by 10:21 - the others. This works remarkably good but you do need to buy or build multiple ladder converters. Holo Audio uses a different approach. According to their documentation they have the microprocessor correct the real world resistor values by using a second bank of resistors to correct the resistor value. It is not clear whether it is done by real time measurement or measured at the production and programmed into the FPGA. The latter is the most obvious and it works extremely fine given the very small deviation of -1.2 dB at a level of -128 dB full scale. As can be seen in this measurement.
Not that his alone 11:04 - makes a good DAC, of course. In the standard mode the May is a non-oversampling DAC. The digital signal is sent to the ladder converter and each bit controls the corresponding switch. An analog reconstruction filter at half the sampling rate then rounds of the samples into an analog signal. The alternative is to have a digital processor upsample the signal and since the sampling rate than changes, a reconstruction filter has to be applied in the digital domain.
These digital filters 11:39 - do cause time smearing. To what degree depends on the power of the processor that does the upsampling and the algorithm use. The idea is that you better have the reconstruction filter in the digital domain do the difficult work making life easier for the analog reconstruction filter. Non oversampling fans will be against it for exactly that reason. The May offers three ways of upsampling: upsampling to a higher sampling frequency in the native format.
12:07 - So DSD is upsampled in DSD format while PCM is upsampled in PCM format. The second option is to upsample both formats to PCM while the third option upsamples all signals to DSD. All four modes can be selected on the front or from the remote control. Switching to and between oversampling modes is instantaneous, switching back to non oversampling does take some time to settle back to the base sample frequency. I suppose this is because the AKM async upsampling chip is no longer in the signal path.
The I²S inputs accept PCM signals 12:44 - up to 1.536MHz, 32 bit and DSD up to DSD1024. USB accepts PCM up to 1.526 MHz, 32 bit and DSD up to DSD512 in DoP and 1024 in native format. SPDIF, TOSlink and AES/EBU go up to PCM 192 kHz 24 bit and DSD64 in DoP. There actually isn’t much to tell about using the May. There is little chance you will run out of inputs or that an input type is not supported.
Switching between inputs 13:29 - is sequential which is less optimal than having direct access to an input by pressing an assigned button on the front or the remote. Especially people - like me - that use a programmable remote to make life easy for the family - prefer direct IR codes for each input. Another inconvenience might be the lack of volume control. For people with an integrated or pre amplifier this is no problem but for those that want to skip the preamp or have active speakers, might miss it. And I hate to have buttons on the remote that are dead.
These small points of 14:05 - criticism are valid to me but will probably be of little consequence for many. There are several ways to fight time smearing in DAC’s, like very long filters, MQA and - as used here - no oversampling. It’s always difficult to discriminate between the approach and the actual design. The May, playing PCM in NOS mode, offers a very high resolution and extremely clean transient behaviour without any trace of sharpness of other nasties. This is clearly of a higher class than my reference, the Mytek Brooklyn Bridge with Syntaxx power supply.
It’s very addicting, so natural and easy on the brain that I am not going 14:54 - to speak of high highs and low lows. It shows a big natural sound stage in which the instrument just are. No over the top focusing or lack of focus. Nothing spectacular, just natural. DSD in non oversampling mode is a tat less impressive but still very good. If you don’t like that, just let me cynically advise you to just switch on any of the oversampling modes to get rid of the very high quality. As far as I can see the oversampling is done by the AKM chip and that is probably too much a consumer level product for this level of DAC.
Luckily there is a very simple solution: just keep it in non oversampling mode. That offers you a very high quality relaxed and natural sound. Not every NOS DAC is also capable of doing native DSD and although I don’t understand how it’s done, the May does it better than my gear. But PCM is where the May is at its best. It goes to show that the power supply plays an enormous part in the quality of the DAC. The May takes that to almost the extreme but it seems to pay off.
The build quality, 16:13 - both mechanically and electronically, is of very high quality. The number of inputs, available in any kind of interface, makes the May extremely versatile. For those that don’t want very high sound quality there even are three oversampling modes to reduce the sound quality. Just kidding of course. Just ignore these modes and you get a fantastic DAC that is worth every penny. That brings us to the end of this video. There will be a new video, as always at Fridays at 5 PM central European time.
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