Connecting your DAC #1:  the interfaces

Connecting your DAC #1: the interfaces

Welcome the The HB Channel, my name is Hans Beekhuyzen and in this show we’ll look at the connection between a digital source and the d/a-converter and what we should know to avoid problems. D/a-converters might have a variety of inputs. Most common are SPDIF and USB, but some come with AES/EBU or I²S. I am often asked what interface is the best. To be honest, there is no definitive answer. All interfaces theoretically work well, it is rather the way they are implemented that makes the difference. The sending device needs to deliver a very clean digital signal, the digital interconnect should not alter the signal in any way and the receiving device should capture the signal perfectly. When this is the case, there will – at least in theory – be no difference between all the interfaces. Unfortunately, it’s not easy and often not cheap to achieve this. There are a number of things you as consumer can do: first choose the best matching source and d/a-converter combination you can find. Read reviews, like on, visit forums, ask friends; in other words, get informed on products. Or visit a trusted dealer and have him demonstrate a good combination. Or do all of the above. Always use a cable that matches the interface you use. A cable that just has the right connectors isn’t necessarily the right cable, as we will see later on. In all cases there are not only wrong and right cables, there are also good cables, excellent cables and bad cables. A good audio dealer will sell you an appropriate cable under the condition you can return it if it’s not satisfactory – undamaged and originally packed, of course. Don’t be fooled by people that say cable technology has no secrets and that all well made cables sound the same. I wish it were. On the other hand, not every expensive or fancy looking cable does sound right. If you want to know at least a number of reasons why both analogue and digital cables can make a great difference: I wrote an entire chapter on this subject in my e-book File Based Audio aka Streaming Audio that is available world wide at Amazon and Apple’s iBook stores. It’s only about 7 euro’s and helps me to produce more videos. Time to look at the digital interfaces, to start with the most popular ones. The AES Three standard, as incorporated in the IEC 60958 standard, comes in four shapes: SPDIF on RCA, SPDIF on Toslink, balanced AES/EBU on XLR and unbalanced AES/EBU on BNC. The data patterns of all four are about the same, but both AES/EBU versions have a slightly different set of metadata, more tailored to professional use. Data wise all four versions can be used on d/a-converters without problems as long as the electrical properties are matched. An isochronous connection is used, meaning that the bits are send at the same speed as they are processed by the converter. The sending device holds the master clock the d/a-converter has to slave to. It is therefore of great importance the bits are sent at a very precise pace, any deviation will potentially produce distortion during the d/a-conversion. The biggest difference between the audio quality of digital sources lie here. Responsible for sending out the audio data at a very precise pace is the designated clock oscillator. Unfortunately that clock signal is easily disturbed by all kinds of interferences. Especially computers, that contain various clocks for various functions, normally are unable to provide a properly clocked digital audio signal. Streaming players and cd-players are small computers too, but well designed ones can produce a very well clocked signal as a result of good and costly engineering. We now have seen what all four AES Three variants have in common, time to look at the differences that mainly has to do with the electrical properties of the interfaces. Two are designed for the consumer market and are known as SPDIF, which stands for Sony Philips Digital InterFace. The first one is an electrical version that uses a rather low 0.5 to 0.6 volt over a 75 ohms coax cable, terminated by RCA or BNC connectors. The construction of RCA connectors prohibit a pure 75 ohms impedance over the required bandwidth, which officially is 128 times the sample frequency. Therefore the use of 75 ohms BNC connectors would be a better choice. These connectors are also used for video and offer 75 ohms impedance over a far greater bandwidth. By the way, there are also BNC connectors and BNC interlinks that have other impedances like 50 ohms, so always check when buying. Unfortunately you’ll usually find RCA’s on consumer equipment. It’s therefore even more important te make sure at least the cable used is 75 ohms. SPDIF can theoretically run lengths up to 10 meters. Some electric SPDIF outputs and inputs are fitted with special transformers that provide galvanic separation to prevent ground loops. We speak of a galvanic separation when there is no electrical connection between one device to another. With digital audio is normally achieved by a transformer where magnetic flux is used to send the signal from the primary winding to the secondary winding of a transformer or by optical cabling as we will see now. The optical version of SPDIF uses a plastic light conductor with either Toslink or Mini-Toslink connectors. It offers less bandwidth than coax cable and is often limited by hardware manufacturers to 96 kilohertz to ensure a reliable connection even under difficult conditions. As mentioned, the advantage of optical is that there is no galvanic connection between the sending and receiving device, eliminating the risk of a ground loop. Furthermore optical connections are not susceptible to radio frequency interference. The optical conductor usually is made of Plastic Optical Fibre, POF for short. Due to optical losses, the maximum usable length is considered to be between 5 to 10 meters. That has to do with the quality of the optical sender, the optical qualities of the cable and the optical receiver. They all can vary in performance and the product of that performance determines the total signal loss. Therefore try to avoid adaptors, from Toslink to Mini Toslink for instance, since any extra connection will create a loss. The more expensive Glass fibre or Silicon fibre cables have a lower loss per meter and therefore distort the digital signal less. That can lead to longer runs and/or lower jitter and they become more affordable by the day. A one meter Toslink to Toslink glass fibre cable can be found on the web at around fifty euros. Balanced AES/EBU is designed for professional use and can run lengths up to one hundred meters due to the 6 volts voltage. It is loosely based on the RS422 serial computer interface and uses a 110 ohm shielded symmetrical cable on XLR connectors. These cables look like microphone cables – as SPDIF cables look like analogue interlinks – but they do need to have the 110 ohms specification. Balanced AES/EBU is also often found on high end consumer equipment, although in most cases it uses only the electrical properties of the AES/EBU standard. The data pattern usually follows the SPDIF standard. For connecting to d/a-converters the SPDIF pattern is no problem while the physical qualities of symmetrical AES/EBU offer more robust connections. Within the professional world the AES/EBU standard has been criticized heavily, for 110 ohm cables were initially hard to find. Furthermore the broadcast industry is used to use 75 ohms coax for high frequency signals. That brought about the second professional version. Unbalanced AES/EBU only differs electrically from balanced AES/EBU. It uses the 75 ohms coax cabling the broadcast industry likes and a 1 to 1.2 volts signal, much like a video signal. AES/EBU unbalanced can run lengths up to a 1000 meters. In fact the physical connection is almost like the electrical SPDIF connection, it only uses a voltage that is twice as high. As far as I know, this interface is not used on consumer equipment. SPDIF on RCA uses 75 ohms cabling, RCA or BNC connectors, is able to do up to 384 kilohertz or higher and is limited to 10 meters length
SPDIF on Toslink and Mini Toslink uses optical cable and is normally limited to 96 kilohertz, in some cases 192 kilohertz, and can run 5 to 10 meters Balanced AES/EBU uses 110 ohm symmetrical cabling, XLR connectors, is officially limited to 192 kilohertz and can run one hundred meters Unbalanced AES/EBU uses 75 ohms cables, BNC connectors, is officially limited to 192 kilohertz and can run 1000 meters All AES 3 interfaces are robust and as long as no a/d or d/a conversion is involved, not very critical. On playback d/a-conversion is involved and since the clock signal is embedded in the data stream, very good cabling and interfacing is eminent if sound quality is your goal. Always use cabling of the right specification and of the highest quality you can afford. Try to avoid ground loops and keep interfering RF signals, like cell phones, wifi and bluetooth at a distance. Doubling the distance between an RF source and the digital interlink reduces the RF stray on the interlink by a factor of four! The USB bus is designed for the computer world and in standard mode is bidirectional. This has the advantage of Universal Plug and Play, UPnP for short, that identifies the device and automatically loads the appropriate driver, when all goes well. Which isn’t always the case with d/a-converters. Sometimes you need to install a proprietary driver first. But let’s start at the beginning. Initially there was an audio-over-USB standard that used isochronous data transport, more or less like SPDIF. This means that the d/a-converter has to slave to the computer’s clock. Since computers have many clocks that can interfere with the audio clock, it will almost never be precise enough to provide quality audio. The USB audio mode does guarantee access to sufficient bandwidth by having the USB interface work in a unidirectional mode. But when the receiving device detects errors, it can’t request a resend. The USB Audio standard was also limited to 48 kilohertz, two channels. When a new standard was released and called USB Audio Profile 2, the old USB Audio Standard was renamed to USB Audio Profile 1 and an asynchronous mode and 96 kilohertz was added. In asynchronous mode the receiver determines the clock frequency and the sender – the computer – frequently asks the receiver what data rate to send on. USB Audio Profile two also has the asynchronous mode and can handle any combination of channels, sample frequencies and bit depths that can be handled by the available total bitrate of the USB connection. Both USB Audio Profile 1 and 2 are supported by all operating systems, including iOS and Android but excluding all Windows versions that only support USB Audio Profile 1. I had hoped that Windows 10 would include Profile 2 support but much to my surprise it doesn’t. It looks like Microsoft does not like audio and video media, they also dropped Windows Media Center in Windows 10 without offering an alternative. Luckily all manufacturers of USB Audio Profile 2 compatible d/a-converters supply drivers for Windows. From 2010 to 2013 a number of d/a-converters were fitted with a proprietary USB protocol that offered an asynchronous data connection and sampling rates up to 192 kilohertz. In these cases support for Linux usually lacks and a proprietary driver is needed for Windows and Mac OS X. Sometimes even different drivers are needed for for specific versions of OS X. In general, when you intend to buy a d/a-converter to use over USB, it’s good practice to check whether it is compatible with the operating system of your computer. Especially older operating systems like Windows XP or OS X 10.6 and earlier, might no longer be supported. If you have a Windows computer, USB Audio Profile 1 works without a driver, USB Audio Profile 2 needs a driver, just like the proprietary async protocols. With Apple OS X no driver is needed for USB Audio Profile 1 and 2 but an OS specific driver is needed for the proprietary async protocols. For Linux: no driver is needed for USB Audio Profile 1 and 2 but a driver for the proprietary async protocols will seldom be available. USB cables will normally comply to the USB standard and have the appropriate 90 ohms impedance. The data channels use voltages up to 3.6 volts, but things are slightly more complex than with the other interfaces. It absolutely pays to use quality USB cables for audio applications. Not only will a quality cable be shielded better, it will also have the correct impedance over a wider spectrum for less distorted square waves. If you want to go geeky: USB also provides a 5 volt power to devices that need it, like thumb drive shaped USB d/a-converters. Usually this power is rather polluted and unless special filtering of that power is applied, it’s better to use another power supply through a special USB to USB connector that takes out the 5 volt power. Some people cut open the USB cable and cut the red power wire or hook it up to a five volt external power. There are many articles on the web on this subject. I²S stands for Inter IC Sound or IIC and is the standard for PCM audio data transport within digital equipment. It was never developed for use between two digital devices and therefore there is no standard connector. For connections between devices the bus uses three lines: the bit clock line, the word select line and a data line. The bit clock line gives one pulse for every audio bit sent, the word clock signals whether the bit sent is either for the left or the right channel. The most elementary connection form consists of three BNC connectors, sending the bit clock, word clock and data line over separate cables. Like the AES Three interfaces it’s an isochronous connection where the source pushes out data at it’s pace and the d/a-converter has to follow. But since a separate clock signal is available, it should work close to ideal, provided the implementation is done well. To prevent timing problems, the three cables should have fully identical properties, including length. Other connectors used are 8P8C – commonly named RJ45 network connectors – sub-D9 – like RS232- DIN and HDMI. I²S works at 2.4 volts but further interface standards are not available since the bus was never intended for use between two devices. The same potential problem as with other busses can occur, square waves can be distorted to a degree that errors occur. But since the clock signal is carried by a separate connection, the system is far more rigid and is considered the preferred interface by many high-end manufacturers. Always assure yourself a matching digital interconnect is available when mixing a source and d/a-converter of different brands. You can read the full article including links on More videos are on the way, so subscribe to this channel, follow my Facebook page or my twitter account if you want to remain informed. You’ll find the information in the description below. Questions can be posted below, on my Facebook of Google+ page or on the contact page on And if you have enjoyed this video, please give it a thumbs up and tell your friends about it. My name is Hans Beekhuyzen for the HB Channel, thank you for watching and see you in the next video or on And whatever you do, enjoy the music!

24 thoughts to “Connecting your DAC #1: the interfaces”

  1. I think you mentioned WiFi close to the system as potentially problematic. Are modern players with WiFi built in addressing this? I had not considered a WiFi effect and will be moving my WiFi router away from my Audio component system tonight. Putting the short cable from wall to router and long cable across the room to the system (rather than the other way 'round, as it is now). Another minute spent with Hans and another measure improved, as always — Lan

  2. Hi,
    I often experience a problem in my portable DAC, Oppo HA-2 connected to iPhone via USB to play music with Onkyo HF player. What happens is that at least 30min of playback, even standing still, suddenly the sound gets all messed up and becomes slow and out of sync. True that the 8cm cable is highly twisted in portable, but as far as I am aware, the clock signal is in the DAC and I never loose the connection. The problem is solved by physically disconnecting the DAC and connect again, or by turning the DAC off and on. What could be the source the problem?

  3. thank you for all information !
    have you any informations about allen & heath zed r16 and how can i use it with an asus notebook (usb 3) !!???

  4. thanks very much for your information, I have a audiolab m dac, could I connect it to my power amplifier with a xlr female to rca lead, instead of just rca, I noticed just using rca, I have to turn the gain way up to get good sound.

  5. i have a separate DAC for my stereo system. it has a toslink and coax input connection. I would like to connect my computer to the DAC. USB is only out for the computer. is there a good quality usb to coax adapter available or am i taking the wrong approach. thanks

  6. im new to all this. is this set up right ? xbox one hdmi direct to tv. and optical from xbox one into dac converter out to 3.5mm jack to pc speakers. i am interested in neoteck 192khz dac digital optical to analog. and avantree dac converter optical to analog

  7. how about connecting my pc to my av receiver via the firewire ilink ports for multichannel high resolution audio

  8. Very good information. Some DVD players now have a decent dac witch make me curious of the HDMI properties do you know the bandwidth limitations of HDMI?

  9. Regarding 13:41 (and as it's up to now still not mentioned e.g. in your Chord Hugo 2 review 😉): Windows 10 has since version 1703 (Creators Update) an inbuilt USB Audio Class 2.0 (UAC2) driver which uses WASAPI instead of ASIO. This availability applies for Windows Insider users from end of August 2016 onwards. That came in handy for me as by coincedence I bought at that time Sony MDR-1ADAC headphones with integrated DAC and just needed to plug it into an USB port of my MacBook Pro which is equipped with Mac OS & Windows 10 (Insider build). So therefore I can confirm that.

  10. when I use optical toslik 24/96 from my computer 15 feet away to my cxn dac/streamer to play flac files recorded by dbpoweramp from cd, the sound is inferior to my cd's 16/44 played thru my gungnir dac. Want to move away from cd's but not if the sound is inferior to cd sourced sound. Did you expect the toslink optical may be my weak link w inferior sound? Perhaps I should replace the optical out from my computer with a $180 schiit eitr( it takes a usb in to coax out, some reclock, gives galvanic separation), or use a network adaptor (lan wire in, usb out to dac) like sotm 200 ultra$1200, or should I run a lan wire to a $120 switch and connect a lan wire into the cxn & use usb out to the dac? Ive seen your setup 1. my goal is to get at least equal sound, comparing a saved flac file recorded from my cd collection compared to cd played thru gungnir dac. is toslink connection perhaps my weakest link, or look elsewhere first.

  11. Hello Sir, I have TV with HDMI port. I want to connect my old 3.5mm jack speakers with TV HDMI port. Pl. Advice

  12. Dear Hans, in the video you mentioned a 50 Euro mark for a good Toslink cable. Is there any such criterion for a SPDIF coax cable other than watching out for 75 Ohm impedance? Specifically I'm looking for a cable connecting the Chord Mojo and the Digi+ Pro, i.e. BNC to 3.5mm mono.

  13. does anyone here know of a way that I can use my iPhone 7 plus as a source to my questyle cma400i amplifier through either optical or coaxial so that I can do away with the issues commonly asociated with usb connections? thank you for any input.

  14. Hi Hans, I have an emotiva XDA-1 DAC, it has a limit of 96 khz, and I wonde if I can update it with a Gustard U12, I don't find many infotmation about this interface, do you know if it will works with my emotiva, and in that case I should connect my cd player or computer direct to my DAC or to Gustard? thanks u in advance 🙂 great videos by the way

  15. Thank You for many things You have taught me, and others have also. With that said, I had never thought of the better option at time being an optical connection; to avoid a ground issue.

    I am working out the "bugs" (and when I say that, I only had a ground loop issue for a moment because I saw what caused it)of a rather modest system. I have about reached what I can afford to get by the "house comity"; don't get me wrong she likes the way things are seen and heard. I will only say that I Thank You, for Your logical explanation of how a person can get the most for a dollar… Thanks again, James

  16. Hi Hans, Any technical explanation why I have a difference in sound if I use Toslink from my PC to DAC vs USB? On Toslink the sound have lower volum and the sound is not clear compared with USB.

  17. Hi Hans, I'm using a MAC with a USB to electrical SPDIF convertor (M2Tech Hiface two) that has its own clock, so it won't be using the inferior clock source of the Mac. I connect this to my DAC that shows the received sample rate. I set the sample rate and bit depth via the "MIDI settings" on the MAC to 24bit/192kHz. So what happens if I play lossless CD content at 16bit/44.1kHz? The DAC still shows 192kHz (or whatever the "MIDI setting" are set to). This must be upsampled right? All my material is lossless FLAC or ALAC at 16/44.1 or 24/96 or 24/192.

    Is the software player doing this? (Plex) or the MAC driver layer or the HiFace convertor?
    Will this affect quality (assuming my cabling and everything else in the chain can recover the 192kHz ok)?

    I have seen software like BitPerfect match the MIDI settings to the sample rate of your source material (which I want to avoid doing manually) for iTunes. Is this a preferred option? Is there a reason to do this?

    Having just watched you video I think the answer might be:
    Upsampling will not affect the quality of the digital signal going to my DAC, but if it re-clocks the signal to a higher bitrate than the source material, you are just making it harder for the DAC to recover the higher rate signal unnecessarily and therefore more susceptible to poor cabling or lower tolerance components in my system.

    Sound about right?

  18. If i have a dac where i can choose if it should use internal clock or use the clock from the sender.

    what would you do?
    i have a Allo usbridge(volumio) with (supra)usb to a smsl vmv d1(dac preamp) to a Nord One NC500DM Stereo connected with (supra)XLR running kimbercables on some jamo c809.

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