In some cases, the AC iPurifier will be used with mains wiring that is balanced, e.g after a balanced transformer, or in some countries, where part of the house wiring is balanced (e.g Taiwan). If the mains wiring is balanced, it will not have a polarity. The AC iPurifier will detect it as 'undefined polarity' and the LED will turn red. However, this does not mean there is a wiring error in-house.
Frequently Asked Questions
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Can the AC iPurifier be plugged into a smart-UPS, battery backup with pure sine wave output or does it need to be plugged into the same wall outlet as my UPS? Answer: The AC iPurifier is for all sine wave AC systems from 100V to 264V AC at 50 & 60Hz. The AC iPurifier cannot distinguish if the source is a generator or a sinewave UPS, so it will work fine.
Yes, it is compliant with CE and all related, relevant harmonised safety standards.
It will not stop it functioning, but signal and speeds may degrade.
Our Groundhog+ also eliminates ground loops, but via a set of connectors different that our GND Defender's IEC interface.
Can you explain that when I use the AC iPurifier in the correct polarity, sound is brighter but in reverse polarity the sound is a lot warmer? All my units are of two-pin except one, which is with earth/ground pin also, due to wiring at my home three-pin version is always in reverse polarity. Why it is happening? Answer: In principle, the AC iPurifier treats live & neutral equal, so it should not make a difference. It could, however, be an interaction with some pieces of equipment that have no earth and an asymmetric internal design. Also, if the units are operated without earth, the common-mode side of the function is not available or functional. You should try to make them all earthed.
It does not use SMPS, it has a special linear power supply unit that works for all mains voltages.
All MOV-based protectors intended for long-term use should have an indicator that the protective components have failed, and this indication must be checked on a regular basis to ensure that protection is still functioning. It's not there, therefore how do we know if it failed? Answer: https://en.wikipedia.org/wiki/Surge_protector#Metal_oxide_varistor This is the source. This indication is not required for standard compliance, only for 'Wikipedia compliance'. That said, the AC iPurifier is fused and in case of a failed MOV it will disconnect from the mains and the indicators will be dark. Thus a 'device failure' indication is implemented, that covers the MOV. Further, the MOV protection in the AC iPurifier (and in power strips etc.) is a secondary (supplementary) protection intended to improve the protection for sensitive and high-value electronics. They are a 'last-ditch' protection in case of a lightning strike or switching transients defeating the primary protection or passing some of the transients. Primary means of surge protection are integrated into mains systems as a matter of standards.
Yes.
My power strip has no ground/earth, but I was actually under the impression that the AC iPurifier established a quasi-ground? Answer: It does not nor can it, it can still reduce differential noise, but without a true earth/ground attached (either via earth connection on the back or from the mains) anything earth/ground related does not work.
Is it impossible to judge the polarity correctly, unless the earth/ground LED turns green? Example: Japan. Answer: Polarity is with respect to earth. Correct polarity has the neutral conductor at (close to) earth potential and the live conductor at the nominal mains voltage (+/- tolerance). Without an earth in the system, it is not possible to determine polarity as it is referenced to earth.
The power bar shown first uses the CEE 7/5 socket (aka 'French', also found in Belgium and Czech Republic). The CEE 7/5 socket has no defined polarity per standard until recently, so it may be wired with the life wire on either side. https://www.toughleads.co.uk/pages/european-sockets Current recommendations are to wire life to the right hole (except Czech Republic which recommends wiring left). If life is wired to the right, the AC iPurifier will indicate correct polarity, because this is the correct polarity. If it is wired in a different way 'the AC iPurifier will correctly indicate incorrect mains polarity, as it is incorrect. However as the CEE7/5 is'polarised', meaning a plug can only be plugged in one way, but the polarity of AC wires is generally undefined, it is not reasonable to expect the polarity indication to be correct unless correct wiring has been verified. Our plug is CEE 7/7 and should correctly engage with the earth pin. If the Earth LED is red, this indicates that no earth connection is being made. Missing earth on a modern mains system intended to have an earth connection is a serious matter and causes serious risks to health and life. If the earth pin in the CEE 7/5 socket is not fully compliant with the standards (diameter too small), it may not make positive contact. In this case it may be necessary to tighten the contact sleeve residing inside the plug after removing the earth connection from the plug, by loosing the screw holding the earth connection part in place and removing it and then pressing the contact sections together. Past that, the AC iPurifiers earth/ground pin is not physically protruding into the ground/earth socket on the international power strip.
This is from one of our former designers, and from an objective, electrical point of view. So please bear with us! https://www.technologyfactory.eu/en/blue-horizon/blue-horizon-mains-noise-analyser/a-5717-320 This does not strictly measure mains noise, but distortion on the mains sine wave in the audio band, which is dominated by low order harmonics. It actually says so in the small print on the case. This kind of measurement device is particularly useless, as the normal rectifier inside equipment will create exactly the same kind of distortion so it must be well filtered anyway to not affect the device's performance. What is more, the'meter'reading is dominated by low frequencies, so it cannot show improvements at higher frequencies! In other words, it does not detect'any change'but on'certain specific changes', which do not apply to the iFi product in question. The AC iPurifier/PowerStation operates only for high frequencies above the audio range, primarily for the noise generated by the now ubiquitous switching power supplies above 10kHz to 10's of MHz and up. For testing, we use a so-called LISN (Line Impedance Stabilisation Network ' e.g.: https://www.com-power.com/lisns.html??) and a 5GHz spectrum analyser with tracking generator (e.g. similar to this: https://www.rohde-schwarz.com/sg/product/fsl-productstartpage_63493-8042.html). Past that the design of an actual electrical distribution system (solid bus bars) is based on minimising added impedance in line with mains and earth connections. Doing so avoids problems caused by circulating ground leakage currents.
I tried a measuring device (power tester) with the AC iPurifier but it seems to be worse than what's stated? Answer: The power tester in question may not in the traditional sense measure noise. In fact, it is nearly completely insensitive to RF noise. Please view the literature (normally on the box), it likely says something along the lines of: 'analyses the distortion of the 50Hz sine-wave', meaning it essentially works mainly at low frequencies. What happens with the AC iPurifier is that our power supplies draws current with a 100 Hz frequency, which the power tester senses as 'distortion', however the way the current is drawn mirrors pretty much any mains powered device. They show no such thing, simply because while these analysers claim to measure noise, they actually do not measure noise. It is worth looking at the small print of what they measure. The way the AC iPurifier is powered is unusual 'it draws current around the zero-crossing of the AC waveform, where normal power-supplies draw no current. This is intentional to avoid any interactions with power supplies in equipment, which generally draw power only on the peak of the mains waveform and non at zero-crossing. However, if a THD analyser is used to'measure noise' the result will be an erroneous reading, as this circuit will create harmonics on the mains frequency depending on the 50Hz impedance of the mains system. The active circuitry in the AC iPurifier targets noise from around 10kHz and upwards to several MHz. Above that passive filtering takes over. A correct test setup would consist of a LISN (e.g. Schwarzbeck NSLK 8127), a CDN (e.g. Teseq CDN M216) and a Spectrum Analyser with tracking RF generator (e.g. Tektronix RSA5103B) to measure the RF Spectrum return loss with and without AC iPurifier between 10kHz & 30MHz (or higher). We use a similar setup with equivalent networks and analyser to evaluate the behaviour of (among other products) the AC iPurifier.
When plugged in the AC iPurifier, the polarity LED shows red systematically. For example,??Hong Kong uses the same plug as UK. Far from being out of phase as the polarity LED indicated. Should I ask an electrician to rewire the Neutral and the Hot? Answer:??AC phase is defined for all systems. As we do not offer a US to UK plug, but we do with EU to UK, therefore, you are likely using some form of adapter. You need to make sure the adapter takes phasing into account. If the phase is wrong it means a wire expected to have no voltage has the full mains and one that is expected to have full mains voltage has non. With some power supply designs, this can cause meaningful differences with ground noise, with others it does not.
Why does my AC iPurifiers polarity LED which is green with the connection, becomes red certain evenings? For example, the LED is green during the day and becomes red in the early evening. Answer: The AC iPurifier polarity LED indicates that there is a significant voltage between neutral & earth, which normally indicates wrong polarity. In this case, it seems to indicate poor bonding of earth and neutral, which may, or may not represent a safety hazard. If the earth remains solidly at '0V' then it merely indicates a high load that shifts the neutral potential far away from '0V' without consequences for electrical safety or the operation of the AC iPurifier.
The surge protection is MOV based. Normally there is no loss of effectiveness unless the surges encountered exceed the rating. If a surge happens the MOV is not toast, it actually resets but with enough excess surge current (e.g. a lightning strike into the line in the wall next to it ' (which is possible in the USA) it would be vaporized. Of course this is an extreme case where there is likely a substantial insurance case and a ruined house. Under normal conditions, everything should be fine.
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There are a few reasons for why the noise readings might increase when using the iFi audio AC iPurifier with a cheap AC noise tester: In summary, inherent Limitations of the Tester: These inexpensive testers are essentially toys and cannot genuinely detect noise. They use transformers with very narrow frequency ranges and detection circuits and components from 30-year-old washing machines. Consequently, they generate their own noise and produce false readings, especially when encountering active noise cancellation like that used in the AC iPurifier. To accurately assess the performance of the iFi Audio AC iPurifier, it would be better to use high-quality, calibrated equipment designed for audio power supply testing. Professional-grade tools can provide more reliable and precise measurements, giving a clearer picture of how effectively the AC iPurifier is reducing noise and improving the power supply quality.
A simple test is to use a multimeter set to AC and 500V AC range and to test the voltage between Earth (black probe) and the two power connections. One should read 0V, the other should read 220/240V AC (depending on the precise local system).
Will the AC iPurifier work with the dual phase power supply? Example: Philippines Answer: The AC iPurifier works with any mains system. For best results, the earth/ground needs to be connected. The polarity indication does not work correctly if earth and neutral are not bonded or the wiring is too think combined with a heavy current draw. This, however, is just indicating a (potential) problem, it does not affect other functionality.
I'm using a local Network based on power-line communication. The slot where I would try the purifier is not equipped with a power-line communication plug but wanted to make sure it won't impact badly my network. Answer: Every situation is different and so the AC iPurifier will reduce noise and data (internet) transferred over mains is one form of noise. The greater the distance between the AC iPurifier and the internet source the better and vice versa. So, you are best to 'try before you buy' by asking some retailer if they offer a satisfaction period.
Answer: The noise reduction system has two components, one that attenuates differential noise (between live and neutral) and one that attenuates common mode noise (between live/neutral & earth). Differential noise reduction (differential noise is noise between live & neutral) remains active even without earth reference. If there is no earth/ground reference (two red LED's) there is no'reference' for the common mode noise reduction circuit. It also means the system operates without mains earth, which in many cases will place the user at the risk of electric shock, including potential lethal shocks. Note that that is unrelated to the AC iPurifier, it merely indicates this problem, it does not cause it. Without a earth/ground reference, the common mode noise reduction circuit is inoperative (common mode noise is noise that exists between the Earth and the two power conductors, Live & Neutral). An earth/ground reference may be provided via the earth/ground terminal on the unit, however this does not replace the safety earth function (though the red LED should turn off). Common mode noise is generally produced by all switched mode power supplies in varying degrees and is the largest problem. It may be possible to achieve substantial noise reduction by adding an earth-wire from the earth port on top of the AC iPurifier to the case of the largest noise producing device (usually the one with the highest power consumption). A better choice is to add a true earth to the system, from earth rods driven directly into the soil (traditional houses) or to approximate one in concrete floored buildings with steel reinforced structures by placing a large metal plate (> 1m^2) on the floor (below tatami/carpet) directly on the concrete. Short Answer: The differential noise reduction remains active even without earth reference, however the common mode noise circuit will be inoperative.
The noise reduction system has two components, one that attenuates differential noise (between live and neutral) and one that attenuates common mode noise (between live/neutral & earth). Differential noise reduction (differential noise is noise between live & neutral) remains active even without earth reference. If there is no earth/ground reference (two red LEDs) there is no 'reference' for the common mode noise reduction circuit. It also means the system operates without mains earth, which in many cases will place the user at the risk of electric shock, including potentially lethal shocks. Note that that is unrelated to the AC iPurifier, it merely indicates this problem, it does not cause it. Without an earth/ground reference, the common mode noise reduction circuit is inoperative (common mode noise is noise that exists between the Earth and the two power conductors, Live & Neutral). An earth/ground reference may be provided via the earth/ground terminal on the unit, however, this does not replace the safety earth function (though the red LED should turn off). Common mode noise is generally produced by all switched-mode power supplies in varying degrees and is the largest problem. It may be possible to achieve substantial noise reduction by adding an earth-wire from the earth port on top of the AC iPurifier to the case of the largest noise-producing device (usually the one with the highest power consumption). A better choice is to add a true earth to the system, from earth rods driven directly into the soil (traditional houses) or to approximate one in concrete-floored buildings with steel reinforced structures by placing a large metal plate (> 1m^2) on the floor (below tatami/carpet) directly on the concrete. Short Answer: The differential noise reduction remains active even without earth reference, however, the common-mode noise circuit will be inoperative.
We have computers/screens etc all in the one extension block, will the AC iPurifier help my audio system against these other devices too? Answer: Digital equipment even if not using SMPS (switch mode power supply) often leaks noise, so yes.
Here is some information regarding the 3D effect inside our products; Link to our papers:??https://downloads.ifi-audio.com/tech-notes/#tn-other
Our products will work with the new Apple Lossless streaming software.
I have an iFi device that is powered by the iPower but might it be possible to use a different power supply that is not dependent on bridge rectifiers (AC to DC)? Answer: At this point, we are unaware of any commercial SMPS that does not include a rectifier bridge, so this requirement is doubtful to be fulfilled by any vendor.
All current iFi products that do not have filter selection use the minimum filter for sample rates up to 192kHz and bit-perfect above 192kHz.
All iFi DSD playing DACs have a -3dB point for around 75kHz / 90kHz (depending on the precise model) if the sample rate is > 96khz. At sample rates lower or equal to 96kHz the sample rate is the main determining factor, with the frequency response being flat to around 0.45 * FS and then rolling rapidly if using the standard filter. The Minimum Phase filter will show around 3dB roll off at 0.45 * FS.
Check out this video towards the end.
5.5*2.1mm is the DC barrel size.
Although this is mechanically possible, one DC Blocker is enough in most cases.
If you'll hear audible hum inside a powered audio product, especially with a toroidal transformer inside, our DC Blocker was designed to either lower or remove that noise.
You don't, it's a passive component that does its job upon connecting it right in-between a power cable and audio product powered by it.
Insert a DC Blocker into the mains input of your equipment at the IEC power inlet. Plug the power cord into the DC Blocker. You can use a DC Blocker on each piece of equipment that has a power transformer in its power supply.
By blocking the DC entering the transformer primary this hum is reduced.
Most of them are, but our DC Blocker is by design as small as possible yet capable of blocking up to 1,200mV of DC voltage.
Yes, DC Blocker is safe to use as it is a fireproof product.
Although this unit will still provide them with power, there is little benefit in doing so. In a switchmode power supply the AC mains is first rectified into DC, after which a high frequency switching regulator circuit takes care of any AC or DC fluctuations on the mains.
You need only one DC Blocker per power bar/conditioner.
Yes, our DC Blocker is a passive power filter indeed, meant to block DC voltage from getting inside an audio component's transformer.
The majority of audio electronic equipment uses a power transformer or transformers for their power supply. Mains power starts out being a pure sine wave at the power station but ends up having a lot of distortion imposed upon it by the time it reaches your equipment at home. The distortion comes from other equipment connected to the same power line and could be industrial or domestic in nature, such as pulse-width modulated electric motors (eg. vacuum cleaners, heat pumps), household heating, washing machines, phase-controlled industrial equipment in factories, light dimmers, etc.
It blocks up to 1.2V of DC on the AC mains from reaching the power transformer.
The DC Blocker aims to reduce noisy physical transformer hum caused by DC on the mains. It's designed for equipment that uses a mains power transformer in its power supply.
The Gemini functions as follows: Audio symbol only carries data & related Power symbol only carries power & related The Mercury is standard having all 4 contained in the one USB cable.
This unit is mainly designed for source equipment and amplifiers with a power transformer rating up to 500W (for 230V) or 350W (for 115V). Using equipment with a higher power rating is ok though, as any more power demanded through the DC Blocker is transferred straight to the equipment. DC blocking will still work, but to a lesser degree.
As toroidal and R-core transformers are more efficient than conventional EI transformers they are quite sensitive to any DC entering them from the mains. If your equipment contains one of these transformers using the DC Blocker may be of benefit. EI transformers will also benefit to a lesser extent.
AMR equipment already includes DC blocker circuitry internally, so there is no need to add this device to them.
Our DC Blocker was designed not to. If anything, it will improve a connected component's sound quality due to removal of excess DC voltage.
The DC iPurifier 2 is rated for 24V which includes normal 5% tolerances, so 25-6v may work, but it will be near its limits. It is possible if the tolerances of the DC iPurifier 2 protection circuitry, and of the 25-6v DC power-supply, accumulate. In a worse-case, the DC iPurifier 2 will enter protection mode and shut down. It may do so after a fair while if the status is just at the edge. So we cannot recommend or guarantee operation with a 25-6v power supply attached.
The provided inverter inverts the wire connections. If you have a centre negative power supply unit and a centre negative device, two inverters are needed. One before DC iPurifier 2 and one after. Setup: Power supply > Centre negative > Inverter > DC iPurifier 2 > Inverter > Device
If you'll hear audible hum on your speakers or headphones, this is due to a ground loop in your setup and our Ground Defender was designed to eliminate this unpleasant noise.
On a fundamental level, if there is enough voltage to make active noise cancellation in the DC iPurifier work (around 4V) it works. Excessive voltage past 24V will activate protection circuitry, as will excessive current or reverse polarity. Otherwise is absolutely agnostic about what it is attached to.
On a fundamental level, if there is enough voltage to make active noise cancellation in the DC iPurifier 2 work (around 4V) it works. Excessive voltage past 24V will activate protection circuitry, as will excessive current or reverse polarity. Otherwise is absolutely agnostic about what it is attached to and consumes around 50mA DC regardless of voltage.
Is it possible to use the DC iPurifier with a splitter connection to feed 2 devices rated at same specifications? Answer: Yes, this is possible, however, it will generate a ground loop that may or may not cause problems in terms of noise or sound quality.
Yes, it works bi-directional and will filter the power in these directions.
Yes, it does contain over-current and over-voltage protection based on thermal fuses, if this protection circuitry is engaged the unit may become very hot and will not produce much output voltage. Therefore, if your unit runs hot it may appear that the circuitry was activated, however, it should not activate until well past 24V DC input. In this case, the unit should have re-set itself to normal operation after cooling down.
Approx 40 minutes with a high-powered charger.
The DC iPurifier 2 draws around 50mA for its own operation regardless of voltage, in most cases this is negligible. The rest of the power is available to the powered device, also any voltage drop is minimal (a small fraction of a volt due to contact resistance).
The NEO iDSD2 is designed with advanced internal voltage regulation and regeneration systems, ensuring consistent, high-quality performance across the 9V-15V supply voltage range. Unlike basic circuits that depend heavily on input voltage, the NEO iDSD2 internally regenerates its voltages, making external supply variations irrelevant to its performance. We understand that audio enthusiasts expect premium performance, and our meticulous engineering ensures a uniform and uncompromised user experience, regardless of the power adapter included.
Orange: Receiving power and powered on. Red (+/-): Receiving the wrong polarity otherwise it will not be lit. For Pro audio people who use the DC iPuriifer 2, a lot of equipment is center negative/outer positive and they can't just connect it in series hence there is an LED to warn of this.
Plug the GND Defender into the IEC socket of your earthed component, then plug the IEC power cable into the GND Defender.
Although these words sound similar, here we are using 'earth' to mean the mains earth, and 'ground' to mean the audio zero volt (0V) line on your equipment.
There are 3 versions;
It is due to different variations and height differences. Variations: There are around 48 different DC barrel types, therefore, each one can not be provided so you may need to contact your device's manufacturer to confirm the specifications and purchase the correct adapter. With 3 DC barrel fittings: 5.5??2.1mm (adapters for 5.5??2.5mm and 3.5??1.35mm incl) the one built into the DC iPurifier 2 is the most common one and along with the two sets of pigtails, covers the majority of devices. Height different: Laptops/routers/streamers vary in height ' so physically, the DC iPurifier 2 may/may not mechanically sit at the power port of the source. It may be worth turning the unit on its side so the height is reduced since the distances from the side to the centre is shorter than the height from bottom to centre by a few MM.
I'm planning to use a DC iPurifier 2 with a power supply that is linear, but unregulated, so it generates some ripple which is about 900mV. Would the DC iPurifier 2 be ready to stop that ripple or at least to diminish it to some degree? Answer: The DC iPurifier is a 'cleanup' device for low levels of noise (a few mV or less) ' 900mV ripple will completely overwhelm its circuitry.
It is 2.7 Ohm, if you see 1.5 Ohm this is a misprint.
I'm curious as to how this GE5670 is offered as compatible with 6922 types. As you know these are quite different, the tubes are not the same? Answer: Yes, they have different pin-outs. However, the actual electrical behaviour is closely similar and well within normal observable variation of the 6922 family. The 2C51 family was the premium version of the 6922 family, but many customers just see the same 9-pin layout and assume they are interchangeable. You know what is involved but again, we do not advise separating the tube from the adapter. Plus the adapter has circuitry to reduce noise which also helps that little bit more. Further question: Even the heater current is slightly different. Answer: Actually, all ratings are with an implied tolerance according to the standards of the time ' some tube datasheets reference the precise EIA standard. Implied for items like heater current is a 20% tolerance. So 350mA vs 365mA is well within normal variations. Additionally, nominally interchangeable 6DJ8 family versions from different (or even the same) manufacturers show a surprising variance.??Our R&D once matched up pairs of NOS Siemens E88CC out of around 50pcs tested (as it so happens) on an Amplitrex. No two tubes had even remotely matching sections and it was impossible to make decent pairs. Many are much further out than a 5670 tested as 6DJ8. Further question: And the operating point and trans-conductance are quite different in comparing the datasheets. Answer: Operating points are examples. They do not mean 'This is the only way the tube must be used!' Yes, 6DJ8/6922 Datasheets use a different operating point to specify performance compared to the 5670. Hence the results are different. In practice, the 5670 works well in almost any circuit designed for 6DJ8 Type valves. We have tested.
The 6922 is rated for 220V Anode Voltage (absolute max): https://frank.pocnet.net/sheets/049/6/6922.pdf The Russian 6N23 which is often sold as '6922' is rated 300V Anode Voltage: http://www.goldenmiddle.com/files/6N23P.pdf The 5670W (which our adapter re-pins for 6922 pinout) is rated for 300V Anode Voltage (absolute max): http://www.r-type.org/pdfs/5670WA.pdf None of these tubes should be used with 400V Anode Voltage. It may work, but it severely exceeds the manufacturer's absolute limits. The 5670 has a higher rated anode voltage than 6922 and matches 6N23. So in any equipment correctly designed to use 6922 valves, it will work fine. In this case, only use manufacturer-supplied valves which have been tested to withstand these above absolute maximum rating conditions.
This is true, however, real tubes also show a significant variance from notional parameters, the match between 6DJ8/6922/ECC88 and 5670/396A/2C51 is close enough that they work even in quite twitchy fixed bias circuits. The only parameter that is of very minor concern is the 5670's 15% higher heater current consumption, (350mA vs 300mA). Actually, various datasheets for the 6DJ8/6922/ECC88 group of valves (including Sino/Sov copies) show heater current ratings of up to 365mA maximum. The 5670 is tightly specified for heater current (more so than the 6DJ8/6922/ECC88 group). Any amplifier/pre-amplifier/DAC using the 6DJ8/6922/ECC88 group tubes should provide enough heater current for the stated maximum and will work fine with the 5670.
The GND Defender won't be of benefit to an unearthed system (where all the components have double insulated 2 pin power supplies). Instead, it will probably help if the system can be connected to mains earth. Ifi audio make the Ground Hog cables for this purpose. Earthing this sort of system may result in less hum and fewer digital errors from digital equipment.
All cables can only degrade. So with cables, you get degradation for impedance mismatch and signal degradation. The closer the cable adheres to the impedance spec (90 Ohm) the less impedance mismatch derived degradation. Equally, the lower the various losses in the cable (related to the amount of copper, plating, isolation, and geometry), the lower signal attenuation, and thus the signal level degradation. If either degradation becomes too large the USB signal is cut or unreliable, before obvious problems are noted (cut out, no connection) sound quality can still be affected. However, the regenerate/reclock/rebalance technology that we implement in our products can largely revert the degraded signal. Good cables simply cause less degradation to start with.
USB was never designed to carry high-quality audio hence in effect laying interconnects and PCs side by side the USB cable was not a 'problem'. Our flagship USB cable, Gemini, separates audio & power (the equivalent of separate interconnects and PCs) whilst the RF silencers on both of these cables are urban dwellers for example, where there are numerous radio stations emitting strong signals that would benefit from. Lastly, the 90 ohms is a big deal as this is USB standards and anything that differs from this may cause some issues. The Gemini functions as follows; The audio symbol only carries data & related Power symbol only carries power & related The Mercury is standard having all 4 contained in the one USB cable.
As there are 2 types of connectors for the Gemini/Merury3.0, here are the EAN numbers so you can inform the retailer for the correct one.
The metal oxide silencers 'de-tune' the Mercury or Gemini. As every cable acts as an 'antennae' then one would want to minimise this. For example in some cities, with phono cables, we have vinyl customers who pick up radio stations on their phono stage. Further question: So basically, one can have as many metal oxide ceramic RF noise silencers in the signal path as they wish since they only help the sound by de-tuning each cable and not affect the signal in any way? Answer: The effect of the filters on the signal is essentially nil. The reason, the signal is a twisted pair in a 100% shield. In order for the filters to impact the signal, they would first have to interact. What the filters deal with is the action of the shield and cable as antennae, to receive radio and other noise. They make the cable a very poor antenna that rejects radio frequency noise strongly.
24AWG and 28AWG.
You can find USB information here: http://www.usb.org/developers/docs/devclass_docs.
So-called 'audiophile USB cables' are often of questionable construction and are neither certified to USB standards nor do they conform to the USB specifications relevant. In this case, they often do not work various combinations due to them straying away from the industry standard. Any cable certified according to and conforming correctly to USB 2.0 standards will work, however.
AMR equipment already includes ground lift circuitry internally so there isn't any further benefit in using this GND Defender with them.
Sound systems can range in scale from being simple to very extensive. Generally the more components that are earthed in the system, the more likely an earth loop will develop. It forms when there is more than one path for the audio ground or shield signal to flow through. For example, if a system consists of a preamplifier and power amplifier connected by RCA interconnects, and both the power amp and preamp are earthed to the mains, there are two paths for the ground signal to flow. One through the shield and return wire in the interconnect, and the other via the mains earth in the power board or house wiring and the two equipment earth wires. Here is where the GND Defender can be of assistance. By inserting it into the earth system of the second (or more) low-resistance earthed component, you may reduce the effect of the earth loop, evidenced by a reduction in hum or buzz
That's correct, our Ground Defender works only with products and power cables based on the IEC standard.
Yes, our GND Defender is safe to use as it is a fireproof product.
The core operation principle of our Ground Defender is the same, but this one was designed to work with AC line.
To maintain earthing safety for connected equipment a maximum of 1.2V is allowed to develop across the 100 ohm resistance in the GND Defender, with excess current being passed through without hindrance.
Ideally you should use one Ground Defender per audio component, but also you should remember to connect at least one such product to your power bar/conditioner directly.
It aims to reduce the effect of ground loops in a system. If more than one audio device in your system is earthed to the mains, you may experience hum and/or buzz. By isolating one of the grounds you may reduce the amount of hum in the system.
The GND Defender is a ground lift device. It inserts resistance between the mains earth and your equipment ground. It is meant to eliminate ground loops in your audio setup, that often lead to audible hum on your speakers.
If anything, our Ground Defender will improve a connected component's sound quality past eliminating a ground loop.
You can control the volume with the volume buttons on the GO bar, you cannot control the volume with your phone or other devices.
The GO bar does not have a battery. It is powered by your phone battery and will remain powered as long as your phone has a charge.
To enhance the user experience during the silent intervals between songs, the device automatically mutes itself upon detecting these silent intervals.
Yes. The battery will charge to 80%. While power is being drawn from the battery with the GO blu switched on, its voltage lowers, and the charging circuit takes this into account. Turning off the GO blu allows the battery to charge fully.
Please unpair all Bluetooth devices from the iOS device and restart. This will clear their cache and enable connectivity without interference.
Apple devices default to AAC at 44.1kHz, standard for AirPlay. In order to transmit Hi-Res via Bluetooth up to 96kHz (limited by the Bluetooth standard) you need to use a Hi-Res Bluetooth transmitter connected via USB to your device, then connect Bluetooth to the Hi-Res transmitter. Alternatively you can connect the GO blu directly to your device with a USB-to-Lightning cable for up to 96kHz capability.
Yes. The 3.5mm output and the positive phase on the 4.4mm output are wired together. However, since the Pentaconn 4.4mm socket is a fully balanced output. it will sound louder than the 3.5mm socket if two pf the same headphones are plugged in.?? Always use a balanced cable with the 4.4mm connection, as it contains the two active negative phases that would otherwise short together on a non-balanced cable.
If a Bluetooth connection has been established, first the connected device will use it preferentially. The USB connection can be selected as an alternative. If you have two devices connected, one with Bluetooth and the other with USB, you can select between them on your two devices by playing one or the other. If no Bluetooth connection is established, and the top LED is flashing blue and red (waiting for a Bluetooth device to pair with), plugging in a USB connection that isn't a charger will shut off Bluetooth.
In pairing mode, shown by the top LED flashing blue and red, the GO blu is searching for a device to pair with. It will go to sleep if one isn't found after 5 minutes. If a Bluetooth device is connected the GO blu will stay on until its battery is flat, or the connected device goes into sleep mode or is turned off. The GO blu then enters pairing mode if another Bluetooth device isn't found and goes to sleep after 5 minutes as above.
That depends on a few things; how loud you play it and the type of music (heavy bass and inefficient headphones use more power), distance away from the connected device, temperature to name a few. But, generally you can listen up to 10 hours or more.
Charge to ~80% by checking the on-screen Charge indicator or Charge Port LED ( see manual for more information on the colour description). Once around 80% remove the charge power cable.
A swelling battery is likely caused by improper use. To prevent a battery from bloating, we suggest making sure that the device isn't overheating and that it is kept in a cool area. Make sure you are using a 5v charger and not a fast charger. We also suggest only charging the device to about 80% and waiting until the battery is about 25% depleted before charging again. If your battery is swelling or you notice it doesn't last as long as it used to, please open a support ticket.
When you long-press the button on the multi-function knob, your device's voice assistant will wake up (if enabled). Eg. Siri on Apple or Google Assistant on Android. The microphone on the GO blu is located next to the charging LED on the bottom. No need to speak right into the microphone, it's quite sensitive.
The iDefender breaks ground/earth loops. It does not 'isolate' in the way that the iGalvanic does. It will, however, stop noise loops through the USB cables and also noise from the USB bus power.
Question: I have FH3 IEM with FIIO LC RD 4.4 balanced cable. I also have Meze 99 Classic with Meze 4.4mm silver cable. So I have all my cables connector as 4.4mm. Now when I listen I hear a hiss (sibilance) issue in many songs (not recording problem) with GO Blu. As IEMATCH 4.4 has high output impedance , is it advisable to purchase it or it may not resolve my problem as it is intended to work with higher end DACs only? Or It will be useful for my setups as well. Please help me with this. Answer:??Yes, IEMatch will work with our portable devices and with our higher end DACs.
Currently, you can only update the firmware from a Windows PC.
Unfortunately, it will not work with Windows Vista.
We have consulted with the Qualcomm team and verified the following information regarding their audio codec technology: The aptX Adaptive codec is inherently designed to offer low latency and high resolution. As a result, it supersedes earlier codecs, namely the aptX LL and aptX HD. Due to the advancements made in aptX Adaptive, these older codecs are now deemed obsolete. Specifically, on contemporary chipsets like the QCC5144, these legacy codecs have not been incorporated, given their redundancy. To clarify, all functionalities once specific to aptX LL and aptX HD are now encompassed within the aptX Adaptive codec. Consequently, they are no longer available as separate selectable options. We recommend the utilization of the aptX Adaptive codec for your needs. We will be making the necessary adjustments to our website to ensure this information is represented accurately.
The Go Pod case has 4 LEDs to indicate the charging level. 1, When the power is within 25 percent, the first light flashes. 2, When the power in 25 to 50 percent of the time, the second light flashes. 3, When the power in 50 percent to 75 percent, the third light flashes. 4, When the power in 75 percent to 100 percent, the fourth light flashes. 5, When fully charged 1, 2, 3, 4 indicator lights are always on and no longer flashing.
The Go Pod has 4 settings that supports an automatic matching function for output impedance: 16 ?? (1 ?? -26 ??), 32 ?? (27 ?? -68 ?? or>1200 ??), 64 ?? (69 ?? -154 ??), and 300 ?? (155 ?? -1200 ??). The settings in parentheses have taken into account the impedance of the wire itself.
The GO pod has volume memory, but you will need to turn on Bluetooth volume synchronization. If you do not turn on Bluetooth volume synchronization the Go Pod will play at the volume level the phone was set to when bluetooth is reconnected. If the phone volume is at max or very high, the Go Pod volume will also be set to max. If you use IEM with the Go Pod, we recommend that you turn on the Bluetooth volume synchronization function and turn down the Bluetooth volume of your phone before using it.
There are certain safety requirements for electrical appliances (including audio) that force either an earthed device (Class 1 equipment) or one with double/reinforced isolation (Class 2 equipment). More here: https://en.wikipedia.org/wiki/Appliance_classes For smaller manufacturers the requirements of testing and approval for Class 2 equipment may be forbidding, so 'boutique audio' often uses Class 1. Either Class is legal and if designed and tested correctly safe. So it is no question of 'should it have an earth'? no matter if devices have earth or not, problems with noise may arise. If a system contains multiple Class 1 devices 'the potential for earth/ground loops is created that can lead to measurable or audible hum and buzz. If a system contains only Class II equipment' the potential for 'missing earth' noise (hum, buzz, RFI etc) is created as now all the audio systems shielding is no longer connected to the local earth but floating, so the shielding becomes an antenna that picks up noise. But if a system contains only one Class 1 device and any number of Class 2 all is well and happy. Due to historical reasons, many people will identify the second case as an earth/ground loop, when in fact it is the opposite! Disclaimer: This article outlines how we would diagnose. Not how you should diagnose. One should always ask a qualified electrician to troubleshoot as mains voltages can kill.
Yes, ground is disconnected 'softly', that is for RF frequencies the grounds remain connected, but for low frequencies (mains frequency and harmonics as well as 1/8kHz USB packet noise) that can cause audible noise in ground-loops if the ground is disconnected.
Until a few years ago much Hi-Fi, AV and computer equipment used Class 2 electrical safety, which included an earth. As a result, often systems had multiple earths, leading to earth-loops with the result that often systems had hum and buzz. As a result, many manufacturers have moved their equipment to what is called Class 3, which is 'double-isolated' and has no earth. All Class 3 equipment contains both intentional (actual components) as well as unintentional (parasitic) capacitance between mains and ground. As long as there is at least one earth in the system this is no issue, as all this noise will return to the earth via this single earth. If there is no earth whatsoever the results are unpredictable and can include hum and other noises picked up via various routes. Adding a supplementary earth for audio use is highly likely to make the system completely silent hence we produced the Groundhog.
Please see the image below, you connect it right onto the DC power supply near the base so it can fit.
Here is a short quick guide:
Question: A ground loop is where there are many pathways to ground in a circuit, right? Answer: This is a basic requirement for a ground loop, yes.
Earth/ground is needed to make shielding against all sorts of noises effective, even if the system doesn't have noises without an earth/ground present it's still recommended. For all mains electrical systems, the earth is a major part and the 'reference' for all circuits. Faults can cause the earth to become something else than zero and that can cause all sorts of troubles.
Battery products: If the DAC/amplifier and the DAP (digital audio player) both run on battery and there are no other connections, then there is no need for an earth/ground connection. Battery products with mains powered device: If any form of mains powered device is connected then the earth/ground connection comes into play, because the mains voltage is bonded and referenced to earth.
All iFi MQA compatible DACs can do the last MQA unfold, thus fully unfold MQA master tracks. MQA renderer DACs do the last unfold and require Tidal etc. to do the first unfolds. MQA decoder DACs can do all unfolds without external unfolding support. iFi MQA decoder DACs include: 'Pro iDSD Signature' NEO iDSD 'iDSD Diablo' ZEN DAC Signature V2 'ZEN DAC V2' hip dac2 Other iFi MQA compatible DACs support MQA renderer only.
We apologise for any inconvenience caused but the hip-dac/hip-dac2 always run off battery power. Please take a look at the user manuals: hip-dac hip-dac2
Question:- Can you explain how it is possible that the DSD1793 can work up to 32 Bit / 786 kHz. The data sheet says it is only 24 Bit / 192 kHz. Answer:- This is sort of our specialty, we have a deep understanding of all the chipsets we used like no other manufacturers, hence we are able to unleash the hidden potential of the chipsets like no other, enabling the DSD1793 to do DSD512/PCM768 is just one of many that we do. That's all we can say'? Bear in mind the datasheet is produced by the Marketing Dept not the R&D of the Burr-Brown AMR has a history of this. We unlocked the DEM (Dynmic Element Matching circuit) in the TDA1541A of the CD-77'that was never mentioned in the datasheet because of the R&D,
We recommend charging the battery to around 50% '75%, not charging to 100%, then using the battery until around 50%' 25% before charging again. Charging should stop when the LED turns from green to white. Keeping the device in a cool dry place will also extend the life of the battery. The battery has a warranty for 1 year, so if you experience any issues, please open a ticket and we'll see what we can do to fix it for you.
Short explanation: An analogue volume control delivers better sound quality than digital volume controls. In case of a small channel imbalance at very low volume settings, please adjust the hip-dac so the volume control is at 12 o'clock for normal listening levels. See below for more detail. Detailed explanation: The hip-dac uses an analogue volume control, specifically a dual-track potentiometer. Being analogue differentiates it from the digital volume control built into DAC chips as it ensures'bit perfect'signal integrity. Digital volume controls alter the original music information (digital data), and often loose effective resolution (bits) even at moderate attenuation; analogue volume controls are free from this defect. One minor downside of using dual-track analogue volume controls is that there is sometimes a slight mechanical mis-tracking between the two channels. This can cause minor channel imbalance at 9 o' clock or lower volume settings, making one channel louder.
The standard, 'b' and 'c' variants of our more recent firmware versions for many devices are ultimately the same firmware, but with some slight variations that may make them more or less suitable for you in various scenarios. Please see below for a more detailed description of each firmware version and when it may or may not be suitable.
The Burr-Brown True Native?? chipset means file formats remain unchanged or 'bit-perfect'. This means you are listening to music as the artist intended in the format in which it was recorded. At iFi we use Burr Brown extensively in our products having selected it for its natural-sounding 'musicality' and True Native architecture. Our experience with this IC means we know how to make the most of it.
We use a??high energy density 4.35V type.
The Hip-Dac is a portable device and it's assumed that it will be moving around in a pocket or bag. Using a a recessed port reduces stress on the cable and the port itself, allowing the device and cables to have a longer lifespan from normal use.
Firstly, it should be stated that ifi's USB driver is a safe and reliable software that will not cause any adverse malfunctions to your computer. There are generally three situations when an antivirus software identifies a virus or gives an alert 1. False alarm problem: It may be caused by incomplete or untimely updates of the identification algorithm of the antivirus software. 2. Unknown file: This may be because the ifi driver file is newly released or uncommon. This can cause antivirus software to fail to recognize it and classify it as a potential threat. 3. Suspicious behavior: This may be caused by the driver's need to interact with the system core and perform sensitive operations. Because it performs actions that are considered suspicious or dangerous, the virus software may flag it as a threat. It is recommended that users add files to the trust list of antivirus software to solve this problem
In the specialist field of audio equipment testing, determining a device's maximum power output is a crucial and difficult task. Many guidelines have been produced over the years to help with this process; these guidelines are the result of different understandings of what makes an accurate and trustworthy measurement of an audio device's capabilities. Music Power, Program Power, Peak Power, Max Power, Instant Peak Power, Dynamic Power, Peak Music Power Output, Total System Power, Instant High Frequency Power (IHF), CAF, UL 1711, IEC 60268-3, EIA RS-490-A, FTC 16 CFR, CEA 490-A, and CEA 2006-B (pertaining to car amplifiers) are some examples of these standards that have changed over time, reflecting the industry's continuous efforts to achieve a more accurate representation of audio performance. Different characteristics of power output and audio quality were the focus of the establishment of each of these standards, which each had its own set of criteria and goals. Others examine sustained performance or particular kinds of audio signals, while still others concentrate on a device's maximum capabilities. These standards all try to give a dependable and uniform way to assess the maximum power output of audio equipment, in spite of their variances. But the intricacy of sound itself makes measuring an audio device's performance precisely difficult. Played through these devices, music is the most prevalent and demanding material. It is a rich tapestry of frequencies, harmonies, and dynamics rather than just a sine wave. Because of its subjective and emotive nature, it is an art form that is difficult to simply quantify. It is therefore necessary to carefully balance the scientific rigour needed for repeatable, objective measurements with an appreciation of the beauty inherent in music in a laboratory test arrangement intended to evaluate a device's performance. Through empirical research and practical experience over the years, we have gained invaluable insights and concluded that using a signal with a single cycle of 1kHz (1ms) repeated at the rate of 100Hz (10ms) is getting quite close to replicate the real-world sensation of listening to music quite accurately. It is defined by a one-millisecond signal that is repeated every ten milliseconds. This particular frequency and pattern of the signal was chosen since it is symbolic of the complexity of music and offers a demanding test of the device's power capacity. Furthermore, for a device to pass this test and be considered capable of delivering high-quality audio, it should maintain less than 1% Total Harmonic Distortion (THD). Additionally, this level of performance should be achieved without the engagement of any protection circuits within the device, which can interfere with the pure assessment of its power capabilities. In summary, the search for a universally accepted standard for determining the highest power output of audio equipment is a continuous process that reflects both the complexity of sound as a medium and our dedication to quality. The goal of this quest is to accurately capture and replicate the spirit of sound, in all its complexity and beauty, rather than just numbers and specs.
We use an AB class amp for all of our products. Please visit this site for more information on AB class amps:??https://www.electronics-tutorials.ws/amplifier/class-ab-amplifier.html
iCan Phantom sensitivity (Vrms): 0dB: SE 11V, Bal: 24V 9dB: SE 4.1V, Bal: 8.2V 18dB: SE 1.35V, Bal: 2.7V
On self-powered devices: The iDefender 3.0 will break the ground as it will interrupt it whilst the voltage bus is high impedance. If the USB device connected is self-powered (it does not draw any power from the USB host), then the iDefender breaks ground-loops by itself. On USB powered devices: The iDefender 3.0 will require a 5v power supply to break the ground or it will be forced to 'bypass' to allow the DAC to work. Therefore 5v is required for the ground to be broken. If we break ground connections without providing separate power source the USB powered DAC wouldn't work. If the USB device draws power from the host this power must be replaced by iPower, or the ground loop cannot be broken, as a low impedance ground is required to draw power.
How much current can the iDefender 3.0 supply when plugged into a USB3.0 port and without an external power supply (?) and max current output without an external power supply? Answer: Drawing power in excess of 5mA will cause the ground isolation of the iDefender 3.0 to be bypassed, it in effect becomes useless. This is primarily a 'fallback' option in case the external supply is lost. If significant power is drawn (> 5mA) please use an external power supply unit.
Yes, this socket is only to charge the iDSD Diablo, which takes ~5 hours with high-powered chargers (such as our iPower X) and ~8 hours with regular chargers.
Question: Using the iDefender 3.0 without external power and connected to a USB-powered DAC it doesn't disconnect the power as it still works? Answer: It contains a 'clapham junction' circuit, which, if the DAC draws power and external power is not connected it switches over to bus power (actually connects power & ground). Question: So you you mean that it only disconnects power when external power supply is connected? Answer: No. It switches grounds (which is what matters) and +5V line to high impedance but leaves them connected (to allow DAC's that require the 5V line for'handshake) UNLESS the DAC draws significant current. In this case it enables a bypass circuit, that is disconnected if external power is applied. So for iDefender without external power: For iDefender with external power: The switch over is transparent and instant. This is aimed at pro audio application where a lost external power would mean a dropped recording, or a dropped set for a DJ.
Our very own OV2028 operational amplifiers inside the iDSD Diablo have THD so vanishingly low (0.00006%) that no volume chip would be able to keep up with them on linearity and distortion. Some regular potentiometers can, however.
Yes, you can, but for the best sound quality avoid charging at the same time.
By taking out the battery, the device will continue pushing power into the board instead of the battery, which will cause the device to malfunction.
Please turn the iDSD Diablo's volume knob until a click is heard and you're good to go.
The iDSD Diablo's approximate battery running time should be around 6 hours dependent upon listening volume and type of headphones.
The iDSD Diablo was designed as a streamlined purist product with sound quality as its top priority. All functionalities that didn't contribute to this fundamental task were purposely omitted.
The iDSD Diablo was designed as a purist device stripped from everything other than essentials that contribute to sound quality. It is a fixed line output.
Yes, which means that the iDSD Diablo has what it takes to comfortably handle pretty much everything from very sensitive IEMs to inefficient planar headphones.
Yes, it is. We're not crazy about digital attenuation and resolution loss that follows.
Yes, the iDSD Diablo is a fair bit above the micro iDSD Signature and also any other of our portable/transportable products or their combination of.
You bet; Panasonic OS-CONs, low ESR inductors by TAIYO YUDEN and Murata, precision MELF resistors, Panasonic ECPU and C0G capacitors, our own OV2028 operational amplifiers and many more.
This 4.4mm output is a balanced line out that allows you to use the iDSD Diablo as a balanced DAC for i.e. active speakers or stereo amplifier.
The iDSD Diablo features batteries but also several of our in-house developed solutions in order to have the power feed to key components be as clean and stable as possible. To combat output impedance inconsistencies in batteries, our step-up transformers operate at extremely high frequency. This allows us to use small, low ESR capacitors that filter easily by fully passive means free from any IC regulators. The DAC chips inside the iDSD Diablo are fronted by extremely low-noise regulators and powered by a headphone amp's pristine power supply. Our Diablo's own 5V USB line is subjected to multi-stage filtering and specialist ultra-high PSRR regulator, whereas yet another of the same type regulates voltage for the product's internal CPU.
The iDSD Diablo connects to an iOS smartphone via a Apple's own USB Camera Adapter, whereas in case of Android devices a regular OTG cable will do.
As with many other iFi audio products, we've incorporated a custom OV Series operational-amplifier. This top-notch component contributes to the extremely low noise, low distortion (0.0001%) and wide bandwidth.
An important aspect of the circuit design is its direct-coupled nature (no coupling capacitor is present), achieved without a conventionally applied DC servo. DirectDrive?'equals a direct signal path which means less signal degradation. This direct signal path is unequalled in audio and is the reason why the sound is so'true'.
Extensive jitter-eradication technologies are applied to the digital stage, including our GMT (Global Master Timing) femto-precision clock and intelligent memory buffer. This represents a total 'out-of-the-box' systematic digital solution that solves jitter once and for all.
The default setting of every Android phone includes a unique Bluetooth codec priority, which we are unable to modify. For example, many phones do not prioritise LDAC by default. The GAIA app has an option to turn off other codecs so that the user can instantly connect to the codec of their choice every time. Note: For phones that support LHDC, it is possible to set LHDC as the default codec without the need for the GAIA app; however, this feature is not available for the LDAC codec.
Both Hi-Res and LE versions of Bluetooth are running at the same time, but only the LE is displayed. This is a small cosmetics bug currently with Qualcomm software, we are working with Qualcomm on a fix on this.
EVERY codec is activated by default when it comes from the factory. It appears that the prior reviewer turned those off for testing because the unit was a review unit.
We cannot comment on the kind of consideration that went into the design of this device. Though given the numbers, it should really come with a dedicated amplifier supplied by the manufacturer, as few commercial amplifiers, headphone amplifier chips or indeed discrete designs can tolerate such a load. Only amplifiers designed for speakers come to mind ' which would usually be unusable due to excessive noise. Question: May we learn more about how R&D came to the 70% minimum impedance value, as well as the 5.6 ohm value? Answer: International standards (voluntary in most countries 'arguably) state that a transducer (Speaker/Headphone/Microphone) impedance rating shall represent the average impedance across the whole 20Hz-20kHz frequency range, however the minimum impedance of the transducer shall be no lower than 70% of the rated impedance, or the rated must be adjusted according to the minimum impedance. This is found in standards and recommendations as far back as the IHF/IEE and the German DIN 45500 standard from the 1960's and is retained in current recommendations. Thus a transducer that has a minimum impedance of 4 ohm anywhere between 20Hz & 20kHz can only be rated as having 5.7 Ohm nominal impedance (e.g. 4Ohm are 70% of 5.7Ohm), even if its average impedance may be 100Ohm. Rating impedance at one specific frequency only, especially if this rating varies greatly from that obtained using the standard method is not covered by standards and in itself neither wrong or correct, however it can be misleading if anyone would mistake this rated impedance as one that is comparable to ratings applied according to the standards. The reason for these strictures is to make sure a reasonable equipment matching is retained. For example, most Headphone amplifiers are designed for headphone impedances between 16 Ohm & 600 Ohm. Higher impedances generally present no problem, but impedance much lower than the intended design target may degrade the objective or subjective performance severely, or indeed may damage connected equipment as excessive current is drawn, beyond design limits. Hence it is important that such specifications are realistic. While designing the iEMatch we applied the same 16 Ohm-600 Ohm target range as reasonable. However we also looked at some of the more extreme common designs (e.g. some from Shure) that have considerably lower minimum impedances than 11.2 Ohm and thus should not be rated as 16 Ohm or higher and actually had one of our researchers comb the entire library of headphone measurements at innerfidelity.com to make sure that we would cover the largest possible range of headphones and IEMs. The manufacturer's site specifies the IEM in question as 12.8 Ohm, which seems in disagreement with the graph included with the comments. With a 12.8 Ohm impedance specification the minimum impedance should be no lower than 9 Ohm according to the standards. Such a specification would be at the absolute extreme of the range the iEMatch is intended to handle and will already be a load few headphone amplifiers can handle well. As it appears the headphone in question is of a much lower impedance (indeed one that is low by the standards of Loudspeakers), it cannot be recommended for use with the iEMatch (though it is unlikely to cause damage), simply because it is far outside the range of impedance the iEMatch is intended to deal with. Furthermore, it would likely be a good idea to make sure that any headphone amplifier it is used with should be explicitly rated for permanent operation into 4 Ohm loads and it should only be used to evaluate headphone amplifiers explicitly stated to be designed for correct operation into 4 Ohm loads. Otherwise the findings made using such an IEM are likely to be very much at divergence with those made making more common and more generally compatible headphones & IEMs. Of course, none of the prior comments should be seen as explicit or implicit criticism of either product or manufacturer. It is up to every manufacturer to decide how they specify their devices and how they design them, even if they decide to not apply best practice or common standards in deriving specifications and if they decide to produce designs that offer poor compatibility with the majority of equipment, well that is their choice. However, it is unreasonable to expect such an extreme product to be compatible with any device that is designed to cover the 99% instead of the 1% and is designed and priced accordingly.
The iEMatch adjusts levels while making sure the source impedance seen by the headphone is low and the load seen by the amplifier is within normal design parameters. It does not alter (materially) impedances, or affect the frequency response of the attached headphones. It is not like some adapters sold that simply place a resistor in series with the headphone and often degrade sound quality.
To make it a bit more clear on our end, iEMatch has at the max under 3 Ohm output impedance. The respective changes in amplitude with frequency generally pretty much parallel the impedance. So with the iEMatch at 1/7th of the output impedance of the above case, the amplitude change with frequency due to this impedance would only be around 1/7th, or less than 0.8dB at the maximum, for this specific example. This should be at the limits of audibility and certainly should not cause large changes in sound quality. There are some possibilities though. First, the parts used in the matching network are Vishay MELFs. These do appear to show quite a pronounced 'burn-in' behaviour and often cause a bright and forward sound in the first few hours. It may be advisable to perform a forced burn-in using the burn-in tracks iFi have made available. There is no need to keep the headphones or IEM's connected for that. Simply select'high sensitivity'and with a headphone attached set a level for playback that does not cause the amplifier to distort, play the burn-in track(s) on repeat for 48' 72 hours and then try to listen again. Please see the included table. It compares the impact of the iEMatch in high and Ultra settings and of an 'impedance adapter' with 33R on the frequency response as well as the respective SPL if using a smartphone as source (approx 1V out) or a serious desktop headphone amplifier (appx. 10V out). Headphone is Shure SE846 based on inner fidelity measurements. As a general rule 3dB is considered the smallest difference in overall loudness that can reliably be identified. Though more recent research places this more at 1dB SPL difference. Similar research suggests that changes in frequency response need to be beyond 1dB if affecting wide ranges of frequencies and between 3-6dB if affecting narrow ranges of frequencies. As can be seen, the 33R impedance adapter would produce gross changes in frequency response, with bass boosted 3dB and the presence region around 5kHz cut by 5dB. This will drastically change the sound character of the headphone. In high sensitivity (which provides a comparable reduction in 1kHz SPL as the 33R adapter), iEMatch limits changes in frequency response to +0.8dB/-1.5dB which may just be at the edge of audibility, but is actually similar to the L/R differences in the measured frequency response of the SE846 by inner fidelity. Using a Smart Phone with appx. 1V output will still raise 115dB Peak SPL, enough to ensure permanent hearing damage with long exposure. In ultra setting the iEMatch will change the frequency response by =0.3dB'-0.8dB, this will challenge the golden of ears to tell. In this case, the SPL from a smartphone at 1V SPL is reduced to 102dB, which may be a little too low for headbanging but will still produce SPL sufficient to damage your hearing with long exposure. All the above applies to only and strictly the Shure SE846, as measured by Innerfidelity, many headphones and IEM will show much less variation in impedance and hence much less variation in frequency response.
We see it quite often that where there is a mix-up of actual, objective dynamic range (that is the difference between a digital maximum level signal and silence) and subjective perception of dynamics. In many modern DAC/headphone amplifier combinations (and the CODEC Chips build into smartphones, DAP's and similar devices) the volume control usually happens in the digital domain, before the DAC chip and headphone amplifier. In this case, each 6dB attenuation in effect lose 1 Bit of resolution and dynamic range. With an attenuation value of 12dB (High Sensitivity) 2 Bits of resolution & dynamic range are restored. With an attenuation value of 24dB (Ultra Sensitivity) 4 Bits of resolution & dynamic range are restored. This is objective dynamic range, as both signal and noise are equally reduced with the signal then being boosted back by reducing the attenuating in the digital domain, ahead of the DAC. The iEMatch by allowing users to increase the digital volume means the recovery of such Bits. Further reading: Now subjective impressions of something sounding'dynamic'are often the result from excessive compression, the polar opposite of'good dynamic range'. Recordings that have good dynamic range rarely sound punchy and'dynamic', except in the loudest passages. For example, try'Jazz Variants'by O-Zone percussion group, this recording has an excellent dynamic range (it holds a DR rating of 16 out of a maximum 20 in the dynamic range database), it does not sound particularly punchy, except when the percussion is hit hard. There are parallels with high-quality TV. A correctly calibrated picture with correct contrast (and high contrast at dark levels) and brightness (and colour) are generally not obviously impressive, the way the'demo mode'on most TV's is. However, it is a more true representation.
The Micro iCan SE is supplied with 15v, however, it has 18v rated protection circuit that kicks in at a little over 18v, but it may not always reset. So it's better not to try anything above 18v.
Question: I was wondering if this was an intended issue with the IEMatch. I have noticed that I am getting some crosstalk even across different headphones/earphones/IEMs and different devices. Doing a left/right test shows that there is some bleeding with the IEMatch that only occurs when it is connected. Answer: IEMatch especially if used in single-ended (and possibly not switched to single-ended) will add some extra resistance to the ground (unavoidable), which increases crosstalk over the situation without IEMatch, but the difference should be modest. Our first reaction would be to check the details, like are you using a 6.3mm adapter? Is balanced or single-ended used? If single-ended, is the unit switched to single-ended? Please open a ticket.
We would not recommend using iEMatch with IEMs that have a real nominal impedance of lower than 12 Ohm. It may work, but such IEMs are problematic, to say the least.
iEMatch gives you the flexibility of 3.5mm TRS or 3.5mm balanced TRRS and a switch between two impedance settings to fine-tune to your IEMs. While the drive impedance to the headphone varies slightly with the two settings, it is a variance between 1 ohm and 3 ohms. This is radically different from so-called impedance adapters, which simply add their rated resistance in series with the headphone. To get the same reduction in hiss and increase of usable dynamic range an 'impedance adapter' used with a 32-ohm nominal headphone as the iEMatch in high sensitivity would require a 100 ohms adapter and likely radically affect the frequency response of this headphone. With iEMatch the headphone drive impedance is 3 ohms which means any change in frequency response is bound to minimal. If we take the rule that minimum impedance >= 70% of rated impedance and maximum impedance up to 300% of rated impedance, the absolute maximum variation in frequency response with IEMatch would be +0.6dB/-0.3dB (barely audible, if at all), while the 'impedance adapter' of 100 ohms would produce +6.1dB/-2.5dB variation in frequency response (grossly audible). So 'impedance adapters' and iEMatch work on very different principles and have very different effects. Therefore any experiences with 'impedance adapters' have no relevance to iEMatch. In other words, 1-3 ohms difference means the iEMatch has the impact of the centre of a donut on the frequency response. One would need a bat-like hearing to tell the difference. We hope this sheds some light as the iEMatch is not a run of the mill product. As always, one should try first.
Yes, the 5v iPower can be used with the Squeezebox Touch. One of our customers has reported the following adapter paired with the iPower is a better fit for the SBT. https://www.switchelectronics.co.uk/catalogsearch/result/?q=210024 However, we have not tried this so it is your decision to follow up or not.
It's designed to be a linear device, which means it's purpose is to get the signal from point A to point B with as little alteration as possible, all the while performing its duty. The??iEMatch adjusts levels while making sure the source impedance seen by the headphone is low and the load seen by the amplifier is within normal design parameters. It does not alter (materially) impedances or affect the frequency response of the attached headphones. It is not like some adapters sold that simply place a resistor in series with the headphone and often degrade sound quality. In fact, the only measurable impedance swing on the iEMatch occurs when you switch it into ultra-sensitive mode. Even in this case, 3 ohms is nothing in the world of headphones. Lastly, it has a measurably linear frequency response. It's not designed to'colour' the sound in any specific way.
Given the wide range of headphones/IEM sensitivities (e.g. from 140dB/V ' a 30dB difference), we felt that as much attenuation as 24dB was needed for the most sensitive headphones. Using a 2-position switch breaks this in halve for less sensitive headphones, so 12/24dB to allow for a compact and affordable product.
The iEmatch is a purely passive device. It cannot by itself produce distortion. Any distortion observed is from the driving amplifier. Further question: But I tested mine with an IEM that's sensitive also, it lowers the volume like an adapter, but it introduces distortion? Answer: It may be that you used a lot of attenuation and then cranked excessively. The amplifier is likely to distort, this is not uncommon.
The iPower does not have a connection to mains earth ' how can it create a ground loop when plugged in directly to mains? Answer: The iPower itself cannot create an earth loop. In fact, it is designed without mains earth explicitly to avoid the potential of creating earth loops! However, as with any power supply, a small amount of leakage current will flow in the power supply unit output (ground/positive). It will be, by design, in the region of 10s of micro-ampere (in other words in compliance with medical equipment directives).
Do the pins have polarity, and if so, does it matter how it is plugged into the outlet? There is no ground pin, so you can pretty much flip it either way. Answer: There is no polarity on any of the iFi power supplies, it does not matter which way around they are plugged.
Items like 'regulation speed' do not apply in the conventional sense, the active noise cancellation system is responsible for dealing with any noise within the audio band, regardless of its source. At higher frequencies where active noise cancellation stops being effective arrays of surface-mounted capacitors keep the internal impedance very low to well > 50MHz. The key regulation limit is actually the resistance and inductance of the cable attached to the iPower, all other parts offer superior performance (as it should be).
Answer: The iPower contains passive filtering on the mains side for EMI purposes (bigger size/value than common). It is quite possible that this helps other devices plugged into the same power strip, though that was not a particular 'design goal'.
It is for both channels separately. This power level cannot be sustained continuously with sine wave testing. It will, however, deliver transients at the rated power without clipping. Therefore 4,000mW 16 Ohm per channel.
Generally speaking, no. It has around 22 times less leakage than a typical SMPS using a similar platform, due to the use of shielding inside the transformer to dramatically reduce the need for the so-called 'Y' capacitor. However, some mains leakage will remain, due to a combination of regulatory requirements on electrical safety and EMC. The mains leakage may be shunted to earth using the 'GroundHog' accessory with any SMPS designed as double insulated (no earth), including but not limited to, the iPower.
First, confirm if the connecting device is centre positive, if not then you will need to use the provided inverter (white adapter) to converter it from centre positive to centre negative. Secondly, with regards to the connecting device you need to match the iPower voltage to match. Third, confirm if the amperage is either higher or lower. Please see the following example. Example: The iPower will work. The iPower won't work but we would insist that you to contact the manufacturer of your device to 100% confirm the specifications before use.
Please see the attached images of the inverter. To properly set this up normally the inverter (white) goes between the iPower DC connector and is then topped with the correct tip which is either supplied with the iPower or you may need to find the correct tip for your product. You can also view our connection diagram here:??https://downloads.ifi-audio.com/portfolio-view/smps-powered_devices/
2.0 metres.
Yes, otherwise it would not pass EMC testing.
The micro iDSD Signature is better than micro iDSD BL in terms of functionality and ease of use.
Question: I have a requirement that a power supply (for medical devices) must be isolated from the main power (galvanic isolation), does the iPower have this? Answer: Our iPower does comply with the medical devices directive, however, there is a small value 'Y' capacitor from the mains to pass EMC, meaning the isolation is not 100%. Indeed, no commercial power supply (linear or switching) will be 100% free from leakage due to basic physics.
As the Pro power supply is a very low noise design with active noise cancellation technology incorporated, replace it with other power supplies (even audiophile linear power supply) will likely degrade the performance of the Pro series, hence it is not recommended.
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Yes, it is RoHS certified. Sometimes it is normal for new power supply to have a small smell upon initial use, after a few days of use, the smell will normally go away.
For the 15v iPower:
Question (example): The noise is given in the audio band. If a 5V power supply is used to power a USB DAC how much significant can be the noise beyond the audio band? Answer: This depends a lot on the design of the DAC itself. Actually, the high-frequency noise in the iPower is also really low. Generally, the noise floor above the audio-band is lower than the audio band noise, which is one of the reasons we only specify audio band noise. There are some noise components from the switching frequency (at around 100kHz). With generic SMPSs & Chargers 0.03V RMS (or 44dB below 5V) noise are common. By comparison in the iPower, this noise is guaranteed to be below 0.00001V RMS (or 114dB below 5V). Indeed, many regulator ICs used for audio have a greater audio band and high-frequency noise than these 0.00001V from the iPower. Hence normally the levels of high-frequency noise remaining in an iPower should be low enough to have no consequence whatsoever on a USB DAC (or any other Audio Device), even one that has only minimal or no built-in power supply filtering/regulation.
Does the iPower have lightning protection built-in case thunderstorms happen or will it affect its performance if plugged into a lightning protector plug? Answer: The iPowers protection is 'normal' ' so do use a protected power strip if your electrical system is vulnerable and you have a lot of thunderstorms.
Based on this (Pi chart)??https://www.raspberrypi.org/help/faqs/ , the absolute max power consumed by the bare baseboard under 'stress' conditions (that normally includes boot) is 1.35A. With screen added this would be 1.82A. If any other peripherals or mass storage devices (e.g. Disks, Solid State memory) are used the power consumption under boot conditions may well exceed 2.5A. As the RPi is a DIY Kit with many configurations it is up to to the person assembling the system to ensure a suitable power budget. If too many peripherals are attached the iPower may be overtaxed during boot, while during normal operation of the basic board with minimal peripherals only a small fraction of the available power is used. Past that the more the iPower has underload the lower the voltage. Open a support ticket with us to re-confirm your setup/system.
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For the iPower, what is needed is an earth/ground on the secondary side to make sure the EMC measures mandated by worldwide EMC Regulations actually work. So for a portable radio without earth/ground it is not a good choice, but for a stationary unit that is earthed/grounded it is an excellent choice.
Question:??Which mA output the noise/ripple figures are measured for the 5V iPower? Answer:??The noise is measured using the closest current load to nominal loading that our dummy load provides. For the 5V unit this is around 1.2A (due to the limits of our dummy load).
We realize that the majority of customers will not use all of the plugs so we have instead reduced the amount due to waste.
That is correct. The iUSB 3.0 can do everything the iPurifier 3 + iUSBPower Micro combination can do and then some. The iPurifier 3.0 + iUSBPower Micro, when set up correctly, delivers a major proportion of the improvement from the iUSB 3.0. If you have the original, we recommend that you only try the iPurifier 3 first with the existing iUSBPower Micro. Don't upgrade to the iUSB3.0 (unless you need two sets of ports and any other additional functionality). For anyone with an existing iUSBPower Micro, adding an iPurifier 3 is a cost-effective way to get a substantial performance upgrade while retaining existing equipment.
The iPurifier 3 is basically 'device agnostic'. It acts as a USB repeater, and is as such, not 'aware' of the signals passing through it. If there are compatibility problems, check if you're using a USB 2.0 or 3.0 port on the PC/laptop. Some older USB 3.0 chip-sets mess up critical timing. If used on a USB 3.0 port, try a USB 2.0 port. If you continue to have issues, please open a support ticket.
To burn in the iTube 2 we suggest connecting it to a DAC run for 24/7 ' this would see proper burn-in period.
A standard 10 metre USB cable will not carry the audio signal properly (it will not work well) but by adding the iPurifier 3, it will actively repeat the signal and extend the run. Therefore, by adding 2 x 5 metre USB cable lengths, then it is definitely possible to use the iPurifier 3 in the centre to extend the USB run. So by adding it to the middle, the chain will look like this: 5m USB cable > iPurifier 3 > 5m USB cable, which results in a 10m USB cable run.
The iPurifier 3 does not affect audio signal levels at all (it cannot).
Yes, the iSilencer draws some extra power as it has active noise cancellation, around 22mA (exact amount depends on power voltage). Further question: Is it possible to use with a smart device like a tablet? Answer: It is hard to say if this will be too much for use with a smart device or alike, though some may deny extra power to be pulled from the device, such as iOS.
The iPurifier 3 will introduce additional latency in the region of a few 100 microseconds or say around 0.3 milliseconds give or take a little. For music production etc. the ASIO or WASAPI driver is the main contributor to latency, seconded by the digital filters in ADC and DAC. The added latency by an iPurifier 3 is minimal by comparison.
The iPurifier 3 draws probably around 120mA as it runs a complete USB 2.0 reclocker/repeater.
You will need to use wired speakers or wired headphones to output audio.
The iPurifier 3 was designed to work at the end of the chain but in the case of the iUSBPower Micro to replicate the performance of an iUSB3.0, then use it before the iUSBPower Micro, so from PC to iUSBPower Micro. Thereafter, use the Gemini from iUSBPower Micro or single USB cable to whatever DAC you have. Reason: The iUSB3.0 is better than the original iUSBPower Micro but unless you wish to have the new extra ports, features etc, we recommend that you try the iPurifier 3 with your existing iUSBPower Micro. The iUSB3.0 will be better, but the iUSBPower Micro is already so good that we cannot ask that you upgrade in the hope of a 'night and day' improvement. Please see the following attachment. Past that, where to place it? (additional)
USB is always bi-directional (works both ways).?? USB defines host and device or in terms of direction upstream (communications from device to host) and downstream (communication from host to device). The host is always on the upstream side and has a type A socket (or OTG cable) the device is always downstream and originally devices were all supposed to be fitted with type B sockets or a type A plug (memorystick etc.). A host (mostly a PC, or android with an OTG cable) is the controlling device and an ADC or DAC is a device. Example: iPurifier 3 plugs into the type B socket on an ADC device and the cable from the computer plugs into the iPurifier 3, exactly as would be the case with a DAC. As long as the iPurifier 3 plugs into a 'B' type socket (or via an adapter into a 'A' plug 'e.g. iDSD micro) it works in the right'direction'.
USB 2.0 is required to offer 100% compatibility with USB 1.0 and 1.1. So yes, all iFi USB products are compatible with USB 1.0.
REclock, REgenerate, REbalance all clean up the signal and restore it to a near-ideal signal (so 'bits are bits'). The hub repeater chip is only the starting point for us. So the reclock is only a part of the equation in the iPurifier 3. The active noise cancellation and power conditioning clean up the power to the USB (if it draws USB power). Additionally, there is a filtering of ground-borne noise (but no IsoGround to deal with low-frequency ground loops).
Our intentions was to create the 4 USB connections however we found that type C and micro USB connection maybe too fragile therefore the type A iPurifier 3 can be used with a short adapter that can convert to type C or Micro.
The iPurifier 3 will mainly help devices that use the isochronous transfer (audio & video in essence), it may (or may not) help with devices that use bulk transfer (hard-drives etc.).
You'll be surprised how stable this product is as it has isolation spikes which can be inserted into the provided ground feet or directly into a carpet-like surface. We've managed to stack approx. 8 micro-sized iFi machines onto an iRack without it toppling over. You build this yourself and can either have 1 shelf or 2 and so forth.
There are 3 versions;
Yes the iSilencer draws some extra power as it has active noise cancellation, around 22mA (exact amount depends on power voltage).
The iSilencer draws a few milliamperes, nothing major. It will get a little warm, that is normal. USB bus power is 5V +/- 10% and should not vary more than that.
The micro iDSD Signature needs about 12 hours to fully charge from a regular USB output, and about 3 hours if connected to a high-powered charger.
As my Micro iCan SE powers a 300 ohm HD800S, would that mean it's operating at class A? Answer: With headphones having an impedance greater than around 100 ohm, the Micro iCan SE always operates in Class A, and thus with constant current draw from the power source, the unit gives off a comfortable warmness. As music tends to have a crest factor (peak to average ratio) of > 10dB, even a signal near clipping on the peaks will have a continuous power that will be 1/10th of the peak power. So in practice and with music, even exceeding Class A on peaks, will have little impact on power consumption from the mains as the bulk of the music remains in Class A.
At least 1 week running 24/7 and as per any normal amplifier it certainly requires an audio signal to be run through it to break-in. We use components from AMR machines which are high-end amplifiers and thus require burning in to sound better.
Gain is the amount of amplification that is applied to the input signal with the volume control set to maximum. More on Gain: https://en.wikipedia.org/wiki/Gain_(electronics) If for example 0dB (amplification factor 1) is selected and the maximum output from the source is 1V, then with the volume control at maximum the output will also be 1V. Depending on the headphone connected the SPL produced by a 1V signal may vary from as low as 85dB to 138dB. Thus gain needs to be considered in combination with the source and the headphone. A headphone that produces 138dB/1V should not be used with even the full 1V signal, while an 85dB/2.83V headphone will need the full maximum output. We recommend setting the gain as low as possible but as high as needed to get comfortable listening levels with the volume control at around 12 o'clock to 2 o'clock.
It may be that the device is not grounded/earthed, so make sure that it is. If done then essentially the problem is most likely cables (could even be headphone cables) acting as antennae and literally receiving the radio signal. We would suggest to take it to another place and re-try it there.
At maximum gain (24dB) and volume set to max, 10V output will require around 0.6V input. Simply set gain to 0dB, then 10V in = 10V out.
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When idle the Micro iCan SE draws around 3 Watt, so 1kWh is consumed every two weeks. The cost of 1kWh varies but in Denmark, it?? isthe highest cost power country in Europe with 30 cent euro per kWh. This means to run the Micro iCan SE for one year non-stop in Denmark would consume around 26kWh and cost under 8 euro.
The maximum output voltage is 10v per channel, for impedances above appx. 50 Ohm. Thus power at any give impedance is 100,000 / Resistance, e.g. 100,000 / 600= 166.6mW or 100,000 / 100 = 1,000mW etc.
We would only add that there are many good Class A headphone amplifiers on the market that sound great but what makes the iCan Micro SE a little above and beyond in a sea of headphone amplifiers is the Tube-State?? circuit which AMR developed originally for the iCan Micro SE. Tube-State'?uses a discrete circuit that deliberately and nicely models the way tubes amplify audio. From our own R&D testing, we can assure you that the Tube-State?? circuitry dishes out quite a large dollop of sonic improvement and is the core ingredient behind its sonics, more so than the pure Class A. As an example; the iCan is nicer sounding than many other desktop headphone amplifiers too (even iFi), because it is deliberately made to sound like a good tube headphone amplifier and so, mains power is needed for Tube-State'?so not possible on battery/USB power at this juncture. Ultimate, Tube-State?? does not match real tubes but it just gets very close.
Generally speaking, electronics are best left running. In 99% of the cases components in iFi products are ran at power/voltage that means there is little difference between shelf life and operation, so this does not really impact longevity either. For devices with tubes, however, there are components (the tubes) which have a limited lifespan and which are not easily or cheaply replaced (NOS tubes are becoming extinct at a rapid rate), so they can (and should) be powered down when not in use to preserve tube life. Past that we adhere to the 2013 EU law on standby.
Yes, they are usable at the same time. Further question: Will there be a decrease in sonics if both are connected? Answer: Very minimal.
By its very functional principle, DSD has RFI of twice the maximum level of the audio signal and four times the nominal DSD '0dB' point between, 1.4MHz and 2.8MHz. DSD must be lowpass filtered to avoid these extremely high levels of RFI going out from the DAC chip and messing with the sound quality. Each filter is different. There is always a trade-off between impulse response, phase response, frequency response, and selectivity (how much-unwanted signals are suppressed). Which filter is preferred depends on many factors and it is hard to predict, simply try them all, pick the one that is preferred.
You can say that the Micro iDAC 2 was actually the first shipping product with active noise cancellation (ANC), it was generally developed for situations where we do not want to drop out any voltage but still want to reduce noise (USB powered DAC is an obvious one) and it was applied for that reason and as an initial case in the iDAC 2. It proved so excellent at the job, that we developed the ANC+ version for super low-noise applications. The Micro iDAC 2 actually has triple cascaded ANC, one directly at the USB input and another on the audio circuitry power supply with the third one at the DAC. The clock, by comparison, has a super quiet regulator, as the clock runs on 3.3V, so we have voltage to spare for a conventional regulator, but our analogue stage and DAC both need > 4.8V for maximum performance and as USB is only 5V, any losses in DC voltage (such as from a traditional regulator) cannot be tolerated.
Further question: Is the Apple device / Android USB audio class 2.0 compatible already? Answer: Yes but see individual FAQ regarding Apple and Android.
The RCAs are fixed output (ie: full volume) whilst the 3.5mm is variable output (ie: as a pre-amplifier/headphone amplifier). Engaging the 3.5mm activates its own powered circuit and the same for the RCAs.
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If used correctly, it'll be good for many years.
Is there a danger for the device if this is happening 4-5 times when using it? In other words, it goes into sleep mode (when I'm not present) and when I return it awakes (repeat this around 5 times per day). Answer: The whole point of the'sleep mode'and'sleep charge'is to maximise both operational battery life and overall battery lifespan. So the only'danger' is that of longer operating time in battery mode and of a longer lifespan of the battery.
The SPDIF out receives the same signal as the main DAC. If DTS or Dolby digital is sent, this is not standard audio, and will cause the DAC to produce loud random noise. The SPDIF port will pass the signal unaltered. The necessary settings are inside the playback software (e.g. for J-River under 'bit streaming'). It is NOT possible to send a separate audio signal to the DAC and DTS or Dolby digital to the SPDIF output.
Short explanation: An analogue volume control delivers better sound quality than digital volume controls. In case of a small channel imbalance at very low volume settings, please adjust the iDSD micro power mode and iEMatch settings so that the volume control is at 12 o'clock for normal listening levels, which avoids using very low settings. See below for more detail. Detailed explanation: The iDSD micro uses an analogue volume control, specifically a dual-track potentiometer. Being analogue differentiates it from the digital volume control built into DAC chips as it ensures'bit-perfect'signal integrity. Digital volume controls alter the original music information (digital data), and often loose effective resolution (bits) even at moderate attenuation; analogue volume controls are free from this defect. One minor downside of using dual-track analogue volume controls is that there is sometimes a slight mechanical mistracking between the two channels. This can cause minor channel imbalance at 9 o'clock or lower volume settings, making one channel louder. The iDSD micro should not be used with the volume set so low, as this not only may subject it to this imbalance, but will also not deliver the best sound quality, due to excessive noise and distortion caused by excessive gain. Form this, the iDSD micro includes both power modes and iEMatch adjustments that make sure any headphone can be correctly matched for optimum volume control setting and sound quality. Please see the manual for these settings, and set the iDSD micro so that normal listening levels are at the volume control around the 12 o'clock position. This ensures not just correct tracking, but also the lowest noise and best sound quality.
I assume the iDSD doesn't do any re-clocking of the SPDIF signal from let's say, an Google Chromecast? Answer: That is incorrect. The input circuitry of the iDSD micro is exactly the same as the stand-alone SPDIF iPurifier. This means SPDIF signals are decoded, send into a memory buffer, de-jittered and then clocked out with the GMT Clock derived from the AMR DP-777.
Connect your phone to the micro iDSD Signature's flat'Type A' USB input with a Lightning to a USB Camera Adapter (Apple) or USB On-The-Go (OTG) cable (Android). When using other audio sources, please connect with a USB cable.
It's bad for lithium batteries to be at 100% all the time. It's better if they do cycles of charging between 20% and 80%. And it's better if you charge them to 60% and then disconnect them completely (you still need to do a charging cycle every few months). For a heavy desktop usage where you intend to have it plugged in forever, we recommend something without battery. Yes, which is why in'desktop mode' (always on) the iDSD micro only charges the battery to 80%, not 100%. For Li-Po batteries there are two things that reduce lifespan, one is staying for a prolonged time at 100% charge, the other being discharged below the minimum discharge voltage. Discharging excessively is prevented by multiple protection circuits and by limiting the charge state to around 80% when the unit is on, the user can choose to avoid excessive battery aging, simply by leaving the unit plugged into an active USB port and leaving it permanently on.
Yes, the excessive temperature will shut down but for that needs to be very hot. Select the appropriate power mode and have good ventilation/air conditioning surrounding the unit.
It does always work because it is a hardware control signal. With the iDSD micro switched on it absolutely can not charge past 4.05V or approx. 80% of full charge. When the iDSD micro is switched off, the maximum charge voltage is 4.2V or 100% Charge. This is controlled by a mechanical power switch, so there is no way for the charger circuit in the iDSD micro to 'disobey.' So when the iDSD micro is switched on, it will 'top-up' charge to 80% in 20 out of 20 top-ups. If the blue LED during sleep is off, it means the battery is charged to 80%, not 100%. The power control system in the iDSD micro is extremely complex, in design and programming we had a state machine model that had 52 different states for the software alone, and there additional controls that are pure hardware and if we include would produce 208 possible states for the power system. However, in use, it is really easy and with a minimum level of care and following instructions, one never needs to know much. Whenever music is playing it does not charge in any way whatsoever, USB power is completely cut off and internally disconnected. Only if no music plays will the iDSD micro will enter sleep mode, where heavy battery consumers are shut off and it will attempt to charge. But iPhones and earlier Android phones (as common when the iDSD micro was released) do not provide large amounts of current (e.g. 500mA) and for those phones, the iDSD micro will immediately terminate the charge attempt (before any protection systems can bomb). It will charge from any attached USB power adaptor or USB port that delivers over ~150mA, which includes modern android smartphones (hence Firmware 5.XB). This is why we coded several flavours of firmware. Long Answer Power Modes 0.1) Battery Power 'turn on iDSD micro before attaching the USB connection. This state is'latched'regardless of connection/disconnection of USB until the iDSD micro is turned off or loses power from a flat battery. In battery power mode, if there is no music playing for 3 minutes, the iDSD micro will go into sleep mode (minimise power consumption) and charge if the host (source) makes sufficient charge power available. Charge is always capped at 80%. 0.2) USB Power' turn on iDSD micro after attaching the USB connection. This state is 'latched' regardless of connection/disconnection of USB until the iDSD micro is turned off or loses power from a flat battery. If the USB host does not provide sufficient power to operate the difference will be drawn from the battery. If there is more power available than needed to operate the iDSD micro, it will be used to charge the battery if the battery requires charging. In USB power mode, if there is no music playing for 15 minutes, the iDSD micro will go into sleep mode (minimise power consumption) to speed up recharging. Charge is always capped at 80%. 0.3) Recharge 'turn off iDSD micro. It will charge from any available power source at the maximum rate supported. Even power sources that the iDSD micro would not draw charge from when turned on will be used. An approx. 18 hour timeout makes sure to avoid overcharging a damaged battery. 0.4) USB BC1.2 standard' For charging use a USB if BC1.2 compliant 'charging USB port.' Many modern PCs and laptops have them, if not BC1.2 charging USB hubs are available. Using BC1.2 ports makes sure that the iDSD micro can draw 1.5 ampere from the USB port, meaning in normal mode it can operate at full tilt and recharge a flat battery within 6 hours simultaneously. Even in Turbo mode the battery will recharge at a significant rate. Not using BC1.2 compliant chargers limit the maximum current drawn from USB to 0.5A. This means in normal mode there is no current 'left over' to charge the battery and in Turbo mode the battery is slowly depleted even if running on USB power. While some non-standard (proprietary 'e.g. Apple, Samsung) chargers may be correctly detected, this is not guaranteed with the original iDSD micro and often Apple (and Samsung) chargers are detected as 0.5A power sources. This is due to Apple's choice to actively avoid complying with industry standards and Samsung following suit. The charger detection chipset in the iDSD micro BL is upgraded from Fairchild to TI which detects Apple chargers correctly 100% of the time. The iFi iUSB 3.0 nano & micro are BC1.2 standard-compliant USB power sources that offer additional benefits for USB audio. 1) Continuous desktop/home use in battery mode 1.1) Preparation 'Make sure to let the iDSD micro charge fully over a > 12 hour period switched off to make sure the battery is at maximum charge. Disconnect iDSD micro and reconnect while switched off. The LED should turn blue for a few seconds, then go off. If the LED stays blue it means the previous charge process timed out before the battery was full, in this case investigate your USB cable and charger/port used, as they may provide insufficient power. 1.2) Operation' Turn on iDSD micro before connecting the USB cable. Leave turned on permanently. Make sure that the USB audio stream cuts off when not using the iDSD micro audio and the iDSD micro goes into sleep/sleep & charge mode. Then, if attached to a BC1.2 USB port and at least given at least 50% 'sleep time' in every 12 hour period running in Turbo, 18 hour period in Normal and 24 hour period in Eco mode, the operation is 'forget about it' and completely transparent and remains in battery mode until switched off or the battery is depleted due to lack of USB power or prolonged use. 2) Continuous use in USB mode 2.1) Preparation 'Make sure to let the iDSD micro charge fully over a > 12 hour period switched off to make sure the battery is at maximum charge. Disconnect iDSD micro and reconnect while switched off. The LED should turn blue for a few seconds, then go off. If the LED stays blue it means the previous charge process timed out before the battery was full, in this case investigate your USB cable and charger/port used, as they may provide insufficient power. 2.2) Operation' Turn on iDSD micro after connecting the USB cable. Leave turned on permanently. Make sure that the USB audio stream cuts off when not using the iDSD micro audio and the iDSD micro goes into sleep mode. Then, if attached to a BC1.2 USB port, operation is 'forget about it' and completely transparent. If attached to a standard USB Port 'forget about it' use is limited to Eco & Normal power modes. 3) Portable use in battery mode 3.1) Preparation 'Make sure to let the iDSD micro charge fully over a > 12 hour period switched off to make sure the battery is at maximum charge. Disconnect iDSD micro and reconnect while switched off. The LED should turn blue for a few seconds, then go off. If the LED stays blue it means the previous charge process timed out before the battery was full, in this case investigate your USB cable and charger/port used, as they may provide insufficient power. If using any recent Android device via USB please change the firmware to Version 5.XB to disable sleep mode. This significantly shortens operational battery life in cases of long periods with the iDSD micro connected but not playing music, however, it avoids drawing a charge from the phone when no music is playing. 3.2) Operation' Turn on iDSD micro before connecting the USB cable. Disconnect and turn off your iDSD micro when not requiring it's used for a long period of time. 3.3) Recharge' Turn off iDSD micro before connecting to a USB charger. Use a BC1.2 compliant charger (iDSD micro BL also Apple or Samsung 1A or higher current Chargers) to shorten charge time. When the LED stops shining blue, the battery is fully charged.
What is the technical reason for volume being always higher when input is from USB versus coaxial SPDIF? Answer: There isn't one. If the iDSD micro BL is fed a'digital full scale'signal, regardless of it being USB or SPDIF, signal levels are 100% identical. If you do have volume difference this suggests that your SPDIF source is not a'direct pass-through', but that some form of volume adjustment and/or other processing happens. It may be needed to adjust a specific settings in the device, driver etc. or simply to make sure the source's volume control (if present) is turned to maximum.
USB via the provided USB 3.0 blue USB cable is our preferred connection. As a general rule all things being equal, the optical connection is not as good as the USB. From the computer to the iDSD micro via USB it handles up to DSD512 and PCM 768kHz, whereas with an optical connection, this is not as assured (it will vary from source to source).
Eco: 2.2v (volume at 100%) Normal or Turbo: 5.5v (volume at 80%) or 6.5v (volume at 100%) It is either 0dB (Eco) or 9dB (Normal/Turbo) Direct: 2v Pre-amplifier: Depends upon the 'mode' please see vol pot instructions.
It is RMS and applies to the headphone output.
Analogue filter is at around 80KHz. So, if the sample rate, allows an 80KHz tone can be output with less than 3dB attenuation. Question: At higher sample rates can it output a 40 KHz signal? Answer: The digital filter will cut off at 0.45 * FS
On USB power: It goes into sleep mode after 15 minutes after playback of music has stopped (all digital inputs). On battery power: It goes into sleep mode after 3 minutes after playback of music has stopped (all digital inputs). It will wake up within 1 second of receiving a valid signal on any given input or of the headphones being connected. Sleep mode powers down the DAC, analogue stages, and the headphone amplifier stage.
The micro iDSD Signature's volume knob powers it on/off.
Blinking Red/Blue 'When the battery is critically low it will shut down the unit to prevent damage to the battery. The LED will blink red to indicate that the battery needs to charge. Please turn the unit off, and restore the battery to full, before turning it back on. While charging, the LED will turn solid blue. Fast Blinking Green/Blue' A green LED indicates the battery is critically low and the battery's own protection circuitry has disconnected the battery to prevent deep discharge that may cause damage to the battery. Please turn off the unit, you should now see a rapidly blinking blue LED. Then unit will now charge slow charge the battery for around 30 minutes. If done correctly, the LED will now be a solid blue and the unit will continue charging the battery at a faster pace. If the unit shuts off while slow charging, you will need to unplug the power cable, and plug it back in. It may take another 30 minutes of slow charging before it switches back to the normal charging mode. If the battery still will not charge, please open a support ticket and we will be happy to help.
There is no fixed 'single' XBass, it is dependent on the model. It is not 'Basshead' type bass boost that boosts bass in an unnatural way, instead, it is meant to compensate low bass deficiencies in headphones. It is best to try with some music that has very low bass. Example: The main theme from 2046 Soundtrack (especially the 'Train Version') or anything with very low organ notes. XBass will not (for example) make a kick/bass drum louder, but it will give it more 'kick' Please also view the spectrum of the iDSD micro XBass with and without. +4dB at 30Hz, around 6dB at 20Hz. This is a substantial boost, but only for very low frequencies (below the kick drum of most rock/pop drumkits. Further reading with regards to XBass:??https://downloads.ifi-audio.com/wp-content/uploads/data/XBass_TotallyAddictedToBass.pdf For some self-assessments you can try to download RMAA (http://audio.rightmark.org/products/rmaa.shtml) and to do a test into any kind of analogue input of a sound-card on a PC/laptop (line-in, iDSD micro in either headphone out or line out with pre-amplifier) and to run the frequency response test.
Yes, you can and yes, it's safe. Please keep in mind that the micro iDSD Signature's battery should get more power than connected headphones consume, otherwise the product will eventually discharge.
No, the micro iDSD Signature can't be used as a power bank.
The micro iDSD Signature's USB-C port is the charge input. A flashing nearby LED indicates charging in progress.
Please set the micro iDSD Signature's iEMatch and Power Mode settings (#12 and #10 in the manual) to'Ultra Sensitive'and'Eco'respectively. Next, please turn the product's volume knob all the way down and connect your IEMs through a 3.5mm-to-6.3mm adapter. During music playback please gradually increase volume level until it's satisfactory.
The micro iDSD Signature's battery will last for about 6 hours if the product is set into'Turbo Mode', whereas'Eco Mode' will double that time. Also connected loads and volume levels factor in.
The micro iDSD Signature features a Pentaconn 4.4mm headphone out, larger LEDs and a sleeker volume knob. Its internal circuitry is optimized and is more direct.
Please check color of a LED located just next to the micro iDSD Signature's charging port. If upon powering the product its charge LED is dark, then its battery is completely depleted, thus please recharge it.
Please adjust the micro iDSD Signature's power mode ('and possibly also iEMatch switch) accordingly to connected headphones/IEMs.
It's entirely up to you. Please try them and if you like what you hear, please feel free to have them turned on.
When connected to the RCA output, the volume control is bypassed.
That's how the micro iDSD Signature's multi-colored LED indicates audio format and sampling frequency of music currently played.
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You can use these outputs to use the micro iDSD Signature as a high-quality DAC.
The micro iDSD Signature is a battery based headamp/DAC combo product fit for transportable and desktop use. It's also the direct successor of micro iDSD BL.
The new Micro iDSD Finale is our latest product in our Micro line of product. The Micro iDSD Finale is an Adorama exclusive product, with only 1,000 units produced. It features the latest 16-core XMOS processor, which supports MQA decoding. The volume potentiometer has also received an upgrade, allowing the Finale to have the best channel balance low volumes, thereby rectifying Left/Right tracking even while the volume is low. We have also upgrade the capacitors to ELNA's high grade capacitor, ELNA Silmic II. Known for its superior acoustic characteristics, the Silmic II uses silk fibre to produce a sound that is both more powerful and mellower. To read a full introduction to the Micro iDSD Finale, please visit the article below: https://downloads.ifi-audio.com/the-finale-feast/
Optical/toslink with iFi units should handle 192kHz, as we test optical inputs with 192kHz. However, just having 192kHz support in the receiver is insufficient. Most toslink transmitters are not able to handle 192kHz and many toslink cables degrade the signal sufficiently to make 192kHz operation unreliable.
DoP can be used over SPDIF (only DSD64 though), however, current iFi products do not support DoP over SPDIF.
The only exception to this rule is if your iLink micro is (and can't be relocated) halfway around a room from the downstream DAC, in which case the jitter would rise due to the distance of the cable run involved.
The driver for the Micro iLink micro makes it a standard Windows audio device, so it supports any output mode supported by Windows and adds ASIO. Computers with Windows 8 and higher will need our driver.
Both Micro iLink outputs are 'super' ' they do have differing levels. Normal output: The normal output is around 0.75V P-P (so within the original SPDIF Standard). High output: The output has a similar level as AES/EBU around 7V P-P. The high-level output delivers more voltage which gives better performance with SPDIF input chips optimised for professional applications (Cirrus Logic CS84XX series and AKM AK41XX series). Tip: The added note would be that that AKM & Cirrus Logic receivers are designed explicitly for balanced AES-EBU connection with the high signal levels. As a result they do not work very well in SPDIF applications. The transformer output of the iLink micro in effect places a balanced signal on an RCA connector so combined with a standard application note input circuit for the SPDIF application, the receiver effectively receives again a balanced signal with AES-EBU levels allowing the receiver optimum operation. The SPDIF receiver on the CS42528 chip appears a derivative of the generic Cirrus Logic SPDIF chips and will likely work better with the high level output and with J.E.T enabled, but please try all options and select that which gives the best sonic results.
15ft (4.5m), for example, is quite long. Well-shielded cables in a 'semi-balanced' configuration should be used. It may also be necessary to look at managing earth/ground connections in the system. Other than these potential pitfalls, even this length of cable will be fine.
If you select a version or RIAA with the rumble filter active then use the selector switch on the front to select Decca or Columbia, is the rumble filter still active? Or is the rumble filter only active in the middle, RIAA position, and only if selected on the back of the unit? Answer: Yes, the low-frequency roll-off is independent of the EQ curve selected.
RIAA cannot be disabled without modifications. We do not recommend DSP/Digital EQ due to the very large amount of EQ needed (40dB) using digital EQ degrades the ADC output by almost 7 Bits. RIAA EQ was designed for analogue systems and is best handled in the analogue domain. The goal is to play 78 records that pre-date the use of EQ during cutting, this is a very specialised application and several small manufacturers offer dedicated solutions for 78 RPM mono records.
No, flat EQ. We find that software EQ loses resolution at a frightening rate. As there is no digital 'boost' beyond 0 dBFS, the digital domain EQ attenuates the midrange by around 4 Bit and treble by 7 Bit. Better to record a signal already EQ'ed to RIAA and dial in the minor adjustments in the digital domain. Recording'flat' and eq-ing digitally is not very productive in the real world.
No, it does not invert polarity in MC or MM.
Input capacitance switch is for MM input only. It is inoperative at MC. Most MM cartridges need a defined (and significant) load capacitance to have good high-frequency extension. Too little capacitance and they sound dull, too much the high frequencies sound peaky. A good explanation of how capacitive loading for MM cartridges works is found here: http://www.hagtech.com/loading.html It also includes a calculator, which must be used with care, as the capacitance of the arm wiring and RCA cable to the pre-amplifier can have significant, but unknown capacitance.
Any mains powered device that has no earth/ground connection will have a small of leakage current on the case (laptops and smartphones for example as well). As the provided iPower powering the iPhono has no earth/ground connection to avoid earth loops, there will a small amount of leakage current on the case, this is normal as the combination (iPower & iPhono) does comply with all worldwide electrical safety standards. Further, once the line out from the iPhono is connected to a normal Hi-Fi system that is earthed / ground (at least one device has a 3-pin plug) this leakage current will be drained to earth/ground. If the Hi-Fi system has no earth/ground connection it is recommended (in general, not specific to the iPower/iPhono combo) to add a supplementary earth/ground link. Should you require one we have produced the GroundHog '??https://downloads.ifi-audio.com/portfolio-view/accessory-groundhog/
If your iPhono is producing a humming noise then please watch the temporary videos as it will explain how to implement an earth/ground into the system. 1)??https://www.dropbox.com/s/6kr156irreqhgut/1.mp4?dl=0 2)??https://www.dropbox.com/s/fruxyw945mklu25/2.mp4?dl=0 3)??https://www.dropbox.com/s/sf45tb2grk71vup/3.mp4?dl=0 4)??https://www.dropbox.com/s/cwzioy3kusuagcy/4.mp4?dl=0 5)??https://www.dropbox.com/home?preview=5.mp4
Reel to reel is too far remote. Phono EQ and tape EQ are very different, even if the standardising organisation is the same and they carry the same name.
It is DC coupled and can 'go off the rails' if there is a large DC input or very high levels of AC (e.g. the 115V/50Hz a 'missing earth/ground' audio system floats at). This 'latches up' the device, it needs to be unplugged for a few minutes and then restarted to be reset. There should be no problem with a properly earthed/grounded audio system, only if there is no earth/ground present. If you are not sure please open a support ticket.
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Please find the provided link which can help shed some light on cartridges. http://www.hagtech.com/loading.html This is a good source to learn from!
Please check our iPhono calculator here: https://downloads.ifi-audio.com/iphono-calculator/
The point of the 'load' is that it is used to optimise the system of records, tonearm, and cartridge. So, it must be ultimately adjusted to what provides the best sound quality. Therefore, there is no 'correct' load that can be set academically, expecting the best results. See what the manufacturer recommends and work from there onwards.
The tube GE5670 can be replaced, yes, it is socketed. GE5670 is a premium of 6922 family so unless Bendix 6N3 or WE396A then not worth rolling. Since tubes are bigger than the chassis size, it is specially designed and positioned on its side. Please see the attached image:??http://www.audiostream.com/images/81413ifi4.jpg As you can see, just on the right of the tube is the black pins. These pull upwards to remove the tube component to replace another. However, this would void the warranty, you are responsible for any damage by trying to open the unit.
Yes, absolutely.
The iTube 2 warms up quickly (under one minute) and sounds best at around the 15-minute mark.
iTube 2 is DC coupled. Meaning there is no coupling capacitor. It uses a DC servo to zero out any DC output on the output to a few millivolts (a few 1,000's of 1 volt). AC is always present when playing music, it is called'signal'.
Yes. The GE5670 is NOS, ie military grade. It is run very conservatively so isn't driven like a normal valve and we are running nominally around 30V/1mA: rated at 100,000 hours. Being designed to run conservatively, you can run it 24/7 for quite a few years if you so wish. Thus 30mW per section, the GE5670 rated maximum is 1400mW per section, so around 2% of rated maximum.
Yes, you can use the iTube 2 in a car, it runs off 12v.
I'm running the iTube 2 as a pre-amplifier but since there is only one set of outputs is there any problem attaching a Y-connector to the outputs so one leg runs to the power amplifier and another to a sub-woofer? Answer: No problem, iTube 2 will drive 600 Ohm loads without trouble.
The actual circuit used is single-ended Class A. What each signature refers to is a harmonic envelope and relative levels of harmonics. These are achieved by adjusting the open-loop gain and thus the amount of negative feedback in the circuit. So the difference between 'classic studio' and 'Push-Pull' is in the levels of harmonics, which are greater for 'Push-Pull' (just as they would be in a Push-Pull tube amplifier vs. a studio pre-amplifier) and feature approx. equal amounts of 2nd Order Harmonics and 3rd Order Harmonics and similar relations between the even/odd harmonics at higher harmonics. For the 'single-ended' signature overall 2nd Order Harmonics is much higher than 3rd Order Harmonics and other harmonics fall off in a similar manner with the even-order harmonics in each pair significantly higher, as would be the case with a low/non-feedback SE amplifier.
Question: What is the maximum RCA input voltage that the iTube 2 can receive in default 0db buffer mode? And in 9db buffer mode? Answer: For undistorted output, the following levels should not be exceeded. 0dB -> +20dBU (7.75V) +9dB -> +11dBU (2.75V) Further question: What is the maximum RCA input voltage that the iTube 2 can receive in pre-amplifier 0db mode? And in 9db pre-amplifier mode? Answer: As there is a volume control that should be set according to the requirements of the source, there is no specific signal level on the input. It is recommended to limit the input to less than 60V AC for reasons of electrical safety. Further question: What is the RCA output voltage in default buffer 0db mode? And in 9db buffer mode? Answer: 0dB output voltage is input voltage. +9dB output voltage is input voltage + 9dB (input *2.81). Further question: What is the maximum RCA output voltage in pre-amplifier 0db mode? And in 9db pre-amplifier mode? Answer: The maximum undistorted output voltage in all cases (pre-amplifier/buffer/0dB/9dB) is greater than +20dBU (7.75V).
It will easily handle the standard 2v input.
150 Ohm. Further question: Does it change when using variable volume control? Answer: No. For a bonus, the input impedance is also fixed, 1 MegaOhm in fixed gain mode and 100k in variable gain mode.
iTube 2 should draw around 0.5A (approx.) @ 12V.
Where you actually want a pre-amplifier, naturally, meaning sources with lower output. Another option when using it as a pre-amplifier for a power amplifier but too much gain without turning the volume down will overload the output stage. Example setup: Device 1: 3V @ 0dBFS. Smartphone: 0.5 ' 1V @ 0dBFS. A gain of 9dB makes such a device similarly loud than a normal line out.
There is no ground/earth.
Question: Is it only and specifically 20Hz that is augmented or is it the bandwidth around 20Hz? In other words, is the Q high (narrow bandwidth) or low at 20Hz when the gain is added? Answer: iTube 2 uses a gentle slope 'shelf' type boost that is at the maximum (12dB) at the stated frequency. It is designed to boost the bass in speakers that have a low reaching but attenuated bass (e.g. sealed box mini-monitors).
Our devices comply with the USB specifications. Interoperability between Thunderbolt and USB needs to be checked with the hardware vendor of the Thunderbolt device. Our understanding is that on Mac, the USB C port can function as either a USB port or as Thunderbolt port, but not both. We would recommend, however, to contact Apple and confirm if this is the case.
Yes, and you would power the micro iUSB3.0 from the 12v of the car lighter using such 12v > DC barrel cables.
My DAC is asynchronous mode and has its own internal master clock, therefore, my question is: who is the master clock in this case? Answer: The micro iUSB 3.0 & nano iUSB3.0 re-clock the USB data stream, not the audio data directly. It improves data integrity and makes sure data is received and decoded correctly. In asynchronous USB audio mode the audio clock is controlled by the DAC and the built-in clock.
As long as the combined impedance of the two headphones is high enough, no issue.
REclock, REgenerate, REbalance all clean up the signal and restore it to a near-ideal signal (so 'bits are bits'). The hub repeater chip is only the starting point for us. So the REclock is only a part of the equation in the USB3.0 (USB2.0 and USB3.0 repeater chips used respectively). The micro iUSB3.0 also has active noise cancellation (ANC) and power conditioning cleaning up the power to the USB (if it draws USB power). Additionally, there is filtering of ground-borne noise (but no IsoGround to deal with low-frequency ground loops).
Absolutely fine to leave on. It is only 9v/2A so very low power, quite green in fact.
Any DC power supply within the range of 7v ' 16v DC with sufficient power (> 12vA) can be used. Further question: Also, if my DAC is not powered by USB, what voltage is recommended for the micro IUSB3.0? Could a 5v or 12v input from the linear power supply be used? Answer: There is no specific voltage recommendation other than that this range should not be exceeded in either direction and it is not related to the fact if power is drawn from USB port(s) or not. Most linear power supplies will tend to have worse noise than the supplied iPower and if using an inadequate linear power supply the performance of the iPower may be worse than using the original power supply.
Yes, it is cleaner than the iPower because both micro & nano iUSB3.0 include double-conversion of the power (which actually makes them quite resilient to what power feeds them) and in actual layout, we implement distinct noise barriers between incoming power and signal section. Each output is then fitted with additional passive LC noise filtering and the active noise cancellation system to remove audio band noise. The result is very low noise, even lower than the iPower itself. We have internally verified this performance previously using the Audio Precision 2 but the measurements at this level are 'challenging' to say the least and we must delineate for the time being, due to lack of this essential commodity.
How is it possible to have a low amperage power supply deliver a higher amperage output via USB? Answer: As the iPower is 9V / 2.0A and the micro iUSB3.0 outputs 5v / 2.5A, the input current is stepped down thus enabling the output (amperage) to be higher. In short, high to low voltage (input) translated into low to high amperage (output).
In order for the micro iUSB3.0 to appear as USB 3.0 hub, the following is required: In this case, the micro iUSB3.0 should appear as USB 3.0. If using with an iFi DAC, they are a USB 2.0 device therefore it will always appear as USB 2.0.
The power LED will be on as long as the external power supply unit is connected and powered. This is normal and intentional. In 'auto' setting the micro iUSB 3.0 will switch off all other circuitry (except the power LED) when the host (PC etc) is off and only the power LED will be on.
The micro iUSB 3.0 is limited by mainly by the power supply that powers it. Using the 9v iPower to operate it, the micro iUSB 3.0 can offer full 1.5A per USB port pair, so 3A in total. Please note that the USB specifications are as follows:
IsoGround: Breaks earth / ground loops. Power mode (On/Auto): Controls if the power from the micro iUSB3.0 is switched off/on together with PC's USB power or is permanently on (e.g. charging a DAC while PC is off). In other words, it continues to supply 5v power to the DAC to keep the handshake alive.
On chipset level we get around 0.5ps (500 femto seconds) random jitter on transmitting. As USB is is bi-directional and has complex data patterns it is non-trivial to measure and specify jitter, also single numbers make limited sense.
Voltage and current rating for the micro iUSB 3.0 are fairly uncritical. The key is that the external power supply must provide enough power (voltage * amperage) to power any devices drawing power from the micro iUSB3.0 and leave some reserve for losses and internal power use (say 1.3 factor for reserves). Voltages from 8v to 16v nominal are fine, an input voltage of 18v should not be exceeded. If we want to power two USB 3.0 devices we need 5v/0.9A X 2 or 9 Watt. Add some reserve and a 12 Watt power supply, which could be for example 12v/1A or 9v/1.5A. This is the minimum needed for USB 3.0 compliance. Internally the micro iUSB 3.0 can handle up to 6 ampere total current on the 5v bus, so for an extreme case, a 40 Watt power supply (12v/3.33A or 9v/4.44A) could be used without issue. However, such high powers are unlikely to be needed in Hi-Fi systems and there is little to no benefit spending a lot extra for higher power ratings.
For any DC powered product such as the SqueezeBox. The iUSB3.0 powers a 5v DC powered device.
Many DACs use XMOS. Very old XMOS firmware versions have known issues with some USB 3.0 systems. Such DACs need a firmware upgrade from the manufacturer 'we at iFi have issued this update years ago. If your external DAC('s) do not work, please go the following route: 1) Use a USB 2.0 on the PC instead of USB 3.0. 2) On Windows check the user has admin rights, if these are missing Windows may prevent the user from adding hardware (the DAC via iUSB 3.0 is considered new hardware by Windows). 3) After adding the iUSB3.0 with a Windows system the driver needs to be re-installed for most DAC's. To be 100% sure uninstall the existing driver followed by a reboot before attempting to reinstall the driver. It also may be required to disable anti-virus software temporarily and/or disable and audiophile optimisers and other similar software. If services have been manually disabled or disabled using optimisation scripts' check that the 'plug and play' service is enabled and running. Other services if disabled may also cause failure to recognise the DAC. If with all these (iUSB3.0 on USB 2.0 port and driver-reinstall etc.) the DAC is not recognised but iFi DACs (and memory sticks, hard drives etc) all work the problem is most likely a firmware issue which must come from the DAC manufacturer. If you are not sure please open a support ticket.
Will the micro iUSB3.0 tolerate both a wider range of voltages? Answer: The micro iUSB3.0 handles anything from at least 6v to absolute max 18v. So it is compatible with notional 12v battery systems for cars (which really are 14.4v and reach even higher voltages under 'load-dump'). You need to, however, feed enough total power to get enough output (no, it does not include an over-unity power system). For two USB 3.0 devices that is at least 10 Watt and to support two BC1.2 devices that is 15 Watt. With the included 9v iPower you can support one USB 3.0 device and one BC1.2 device.
You can run and power the device straight from your PC taking from the USB power line, however you must connect the unit then switch on, whilst the on board battery can power the unit when your out and about etc by simply switching on first and connect to the PC/smart phone.
The battery inside these units are long lasting and you'll easy reach 4+ years life span. Battery life is normally around 500 full cycles (full charge and run flat), but that means there is still 70% battery capacity left. If less than full cycles, then more partial cycles are possible. If (say) you do a cycle from 100% charge to 50% charge, this is half a cycle. So you can expect to do 1000'half cycles'. Of course, temperature is also a factor.
Yes of course, this can be used as a pre-amplifier. You just control the volume. If you want to bypass it then put the volume on 100% (max).
Yes you can but it is only SPDIF coaxial out. Past that, the iDSD nano LE has no digital output option.
Yes but the iDSD nano is only 2 channel therefore your A/V system must be swapped over to 2 channel instead of multi-channel.
Question: It states that the coaxial digital output is PCM 192kHz, is this PCM 24 bit or is it variable depending on source from 16 to 32bit? Answer: SPDIF only transfers up to 24/192kHz ,so if you see 32 bit, that may be difficult as it's based upon the SPDIF protocol. Question: If the source is 24 bit is it down-sampled or kept at 24 bit? Answer: Up to 24/192kHz it keeps it original. Question: Is the digital coaxial output bit-perfect compared to the USB PCM source? As in the signal is not going from digital > analogue -> digital, instead USB digital > digital SPDIF. Answer: Yes, bit-perfect as the Burr-Brown does not perform internal conversion otherwise it would defeat the object. Question: The iDSD nano can work with all types of inputs (PCM, DSD, DXD) using USB port but on SPDIF coaxial output it will be only PCM and what will be with DSD/DXD? Would they will be convert to PCM as well? Answer: SPDIF only transmits PCM. Hence, you have to setup the software to convert. This is related to the technology/protocol of SPDIF that it only transmits PCM. The transmitter/receiver technology is simply not available.
The RCA's output 1.65v. This is lower than the industry line level output of 2v.
DC offset is normally a few milli-volt.
We use industry standard of 0dBFS = Digital full scale or maximum undistorted signal and we generally have 2v nominal output. Extra: What signal level is used as '0(dB)VU' is arbitrary. When we was still working in the recording side, East Germany used -14dBFS as 0VU, other studios used -18dBFS or other values. Nowadays this usually called LUFS (Loudness units relative to full scale) and is calculated according to EBU recommendation 128. This is a bit more sophisticated than 'VU' Music streaming services like Youtube etc. tend to normalize music to around -15dBFS = 0VU ( -13 dBFS for YouTube, -16 for iTunes and Pandora, -14 for Spotify and TIDAL). This definition does not really relate to equipment in itself, just to the programme material. VU stands for 'Volume Unit' (though more cynical sound engineers like myself called it 'Virtually Useless' even in the 80's and strictly used PPM [Peak Programme Meter] for recording and mastering) and is a fairly long-term average of signal, so it represents more or less well how'loud'a track sounds. Many mastering engineers commonly break any 0VU/LUFS rules and (re)master music as loud as possible. So some tracks may actually have -6dBFS as 0VU, before being normalized. An extreme example, Guns'n Roses 'Welcome to the Jungle' is -2 LUFS'
The iDSD nano has an analogue stage bandwidth of 7Hz to 75kHz. High-frequency cut-off will depend on the sample rate played (192kHz or higher), material at quad speed and above will take advantage of the full analogue bandwidth.
No max but minimum is 12 Ohm.
The small clicking sound is due to the fact the iDSD nano is using a 'stepper' analogue volume control for the best sound, hence while changing up and down the volume, there will be small clicks when volume changes from one step to another, and then to another and so on. The extent of the clicking varies a little bit between different units of iDSD nano.
With 4-pin 3.5mm plugs on smartphone headsets it cannot be guaranteed that the ground makes reliably the correct contact with sockets designed for 3 in 3.5mm plugs TRS. Therefore we would recommend to use 3-pin plugs and if necessary a suitable adapter.
The iOne nano is really the product focused on desktop/home use, but the iDSD nano BL will be fine as well.
Do not use poor quality cables, especially cables with a thin conductors and high resistance as it can lead to problems. Please use the provided USB 3.0 cable or cable directly from a reputable manufacturer.
The nano iDSD BL receives digital, only. Most iPods do analogue out, and there are only a certain generation of iPods that can do digital out. You can find out if your iPod has a digital output via this site: https://files.freemyipod.org/~user890104/bootloader-ipodclassic.html Just check the model number on your iPod. If your iPod can do digital out, you may need to 'jail break' via 'Rockbox'. However, we can't advise this as it could brick your device. If you wish to try, it would be at your own risk. See link here: https://www.rockbox.org If you do this, you must make sure it is an official Apple Camera Connection Kit. See link here: https://www.apple.com/uk/shop/product/MC531ZM/A/apple-ipad-camera-connection-kit
Yes, the nano iDSD BL can still be used in USB mode even when the battery is flat. Although the battery will not be charged whilst it is being used.
No, the nano iDSD BL does not have a sleep mode.
For high impedance headphones or low impedance headphones at low signal levels ' Yes.
Yes, it is possible. If you power it up in USB mode, it will charge the battery and then simply remain on continuous. If you power it up in battery mode, it will deplete the battery and then start pulling power from the USB connection but not charge the battery.
Yes.
The NEO iDSD works with all chargers within specs (5V/2.5A), however, for maximized performance it should be used with our ultra-quiet iPower/iPower X/iPower Elite.
Question: With iDSD nano BL, the description says '2 track potentiometer'. Do I get it right that in iDSD nano BL this is a classical in-signal-path variable resistor that regulates the volume and if so do I get it right that '2 track' means separate tracks for right and left channels to support the 'S-balanced' topology? With the original iDSD nano you called 'software-controlled analogue volume control', meaning that the actual potentiometer that you turn with the knob is not in the signal path, but instead it controls the IC that switches the signal between many different pairs of resistors? Answer: That is correct. After extensive testing we feel the solution in the iDSD nano BL sounds better than in the original iDSD nano.
It does, it just depends on the headphone/volume level and direct/iEMatch mode. Here are some of our tests using Sennheiser HD201
You will need to check USB suspension settings as we suspect the USB ports are still powered when off, for charging. Normally there is a Bios setting to deal with this.
The guaranteed maximum downstream current provision complies with the USB 3.0 standards, that is 5v DC @ +/-0.25v unloaded, with 900mA maximum current at > 4.5v DC.
Most USB 2.0 ports support > 0.5A in practice. A key factor is the USB cable, excessive voltage drop due to high resistance will cause problems and excessive current draw. Use the included USB 3.0 cable for testing. We did a test and needed a very 'heavy' USB device to come close to 0.8A. Using a more typical USB DAC (iDAC 2) we get around 550mA.
We have found out that some USB DAC's do not comply to USB standards. For example, there is one customer where it did not work with a special USB card, why? Because as they had it installed, it output 4.5v not 5v power on the USB bus, which is not to specification compliant (5v +/-5%), in this case operation cannot be guaranteed. Equally, if one or both sides of the system float without earth connection it may be necessary to set the earth link switch correctly to ensure reliable (or any) connection. The recommended sequence is: 1) Use known good cables or correct specification. Generally speaking, those included with iFi products and / or good grade generics will be correct, many audiophile cables are not correct in terms of specification. 2) Make all connections to iGalvanic 3.0 and set the earth link switch as required. Make sure that the port chosen on the source is a fully USB standard compliant port. 3) Make the connection to the source (PC, Mac etc.). Remember that USB negotiations and enabling/updating drivers etc. may take up 30 seconds + before a new device is operational after connection. 4) In most operating systems the attached DAC will show up as a new device, so it may be necessary to re-set the audio configuration of the machine, of playback software etc.
USB signal integrity contains a number of factors, jitter is one, signal rise-time, signal level and signal balance between the two wires are other factors and excessive ripple from reflection within the cable is yet another factor to impact signal integrity, potentially enough for data corruption (which is not corrected in isochronous streaming mode). Impacts on signal integrity are produced by among others (an exhaustive list would be challenging) by cable impedance mismatch, impedance mismatch at connectors, electrical fields from nearby cables and transmitters and not the least isolation devices. In iFi's USB products that incorporate the Reclock / Regenerate / Rebalance technology all of these factors are addressed comprehensively and signal integrity is restored to'close to ideal'.
The vast majority of full-sized dynamic and planar-magnetic headphones will be very happy with the NEO iDSD, even those considered as notoriously hard to drive.
If the switch needs to be set to provide either RF grounding or grounding down to DC, the most likely reason is that there is no ground/earth for the system on one or both sides of the iGalvanic 3.0. In this case USB cables are in effect not shielded against ground/earth referenced noise sources and may therefore result in dropouts if interference is present. Where one or both sides of a system have no ground/earth connection the earth connection should be restored, for example using iFi's'Groundhog' or by switching the ground link switch on the iGalvanic 3.0 correctly.
Let us clarify: 1) Light switches triggering, this means it is the spark energy when switching. Early radio used 'spark gap' transmitters to transmit Morse wireless, the same physics are at work here. 2) The problem is that the spark impulses use the house wiring as antennae transmitting the signal. 3) On the Audio side all and any cables become an antennae receiving this 'blip' signal. If a wire is shielded and the shield is grounded/earthed, the 'blip' should be rejected. The probable cause here is either poor shielding or a lack of grounding or both. It is unlikely that adding any form of isolation will help, as this will not help the shielding/grounding. It may be 'unsolvable' if the house has a poor earth connection (we did experience this in one apartment our colleague had). The best solution is to defeat the actual spark. This may be done by having a competent electrician replace the mechanical relays/switches with modern electronic alternatives, or by applying suitable 'snubber' capacitors across the switch to solve the problem at the source. However, it is possible we are wrong but its a good starting point.
This is normal, if you draw more current downstream heat will increase.
The iGalvanic 3.0 is USB 3.0, so if using USB 3.0 then that is 5GB/sec.
Yes, if you draw more current than iGalvanic 3.0 or the source can supply, the power supply in the iGalvanic 3.0 will limit current by lowering the output voltage.
Is the USB device self-powered? If so it could be because its powered externally and when booting up the computer or source there is a loss of connection with the iGalvanic which the iGalvanic detects the power line first. It normally starts with USB power being applied. If the DAC or USB device is powered externally and already powered before source/PC and iGalvanic are powered, the DAC or USB device may have already timed out the USB connection before everything else is working. Try powering up the PC/source first then connect the iGalvanic to the computer, then connect USB to USB device then power up the self-powered device, so hopefully, the detection is made.
Bluetooth has latency in the region of 200mS.
Many DACs use XMOS. Very old XMOS firmware versions have known issues with some USB 3.0 systems. Such DACs need a firmware upgrade from the manufacturer 'we at iFi have issued this update years ago. If DACs do not work, please go the following route: 1) Use a USB 2.0 on the PC instead of USB 3.0. 2) On Windows check that user has admin rights, if these are missing Windows may prevent the user from adding hardware (the DAC via iUSB 3.0 is considered new hardware by Windows). 3) After adding the iUSB3.0 with a Windows system the driver needs to be re-installed for most DAC's. To be 100% sure uninstall the existing driver followed by a reboot before attempting to reinstall the driver. It also may be required to disable anti-virus software temporarily and/or disable and audiophile optimisers and other similar software. If services have been manually disabled or disabled using optimisation scripts 'check that the'Plug and Play' service is enabled and running. Other services if disabled may also cause failure to recognise the DAC. If with all these (iUSB3.0 on USB 2.0 port and driver-reinstall etc.) the DAC is not recognised but iFi DACs (and memory sticks, hard drives etc) all work the problem is most likely a firmware issue. DAC firmware updates must come from the DAC manufacturer. If you are not sure please open a support ticket.
Yes the iUSB3.0 handles audio and power according to USB Audio Class 2.0 & 3.0. Therefore, it passes all audio types (DSD, DXD, PCM) and files through just fine.
Yes, you can. The NEO iDSD will send audio signal to them simultaneously.
Yes, the NEO iDSD will work with IEMs.
Yes, it is cleaner than the iPower because: The iUSB 3.0 nano and micro include double-conversion of the power (which actually makes them quite resilient to what power feeds them) and in actual layout, we implement distinct noise barriers between incoming power and signal section. Each output is then fitted with additional passive LC noise filtering and the ANC (active noise cancellation) system to remove audio band noise. The result is very low noise, even lower than the iPower itself. We have internally verified this performance previously using the Audio Precision 2 but the measurements at this level are 'challenging' to say the least and we must delineate for the time being, due to lack of this essential commodity.
In short, yes, it will still help as it Reclocks, Rebalance, Regenerates the USB data stream. However, it maybe overkill as the iPurifier 3 also has these features so we would recommend this before the iUSB3.0.
Absolutely fine to leave on ' only 9v/2A so very low power, quite green in fact.
For any DC powered product such as the SqueezeBox. The iUSB3.0 powers a DC powered device.
The NEO iDSD can comfortably work with minimal space around it. However, it should have some freedom around just as any electronic audio device.
Press and hold the knob (3 seconds) to enter brightness selection. Rotate the knob to cycle through the three (high/low/off) brightness modes and press to accept when you get to 'off'.
Yes, you can, however, this will limit total power output delivered to each headphone socket as both rely on the same amplifier circuit.
Yes, as long as this USB socket can provide stable 5V/0.9A. However, regular USB ports are noisy, which might audibly impact the NEO iDSD's performance.
Yes. While constantly pressing the NEO iDSD's rotary knob, please turn on the product to enter the analogue output setting selection menu. The NEO iDSD will remember a previous setting.
Just like any other iFi audio DAC, our NEO iDSD's USB input also features a number of our proprietary measures to sound the best it possibly can.
You press and hold the knob (3 seconds) to enter brightness selection. Rotate the knob to cycle through the three (high/low/off) brightness modes and press to accept.
The MQA logo will be displayed on the NEO iDSD's OLED screen.
One press of the NEO iDSD's rotary knob is all it takes.
No, the NEO iDSD uses the newest 16 Core 2000MIPS XMOS low latency microcontroller. It has the capacity to be able to play both MQA and DSD512 right out of the box without the need for firmware changes.
The NEO iDSD relies on a USB3.0 socket. Although it will work perfectly fine with regular (USB2.0) cables, a USB3.0 cable such as iFi's Mercury/Gemini are advised for reliable connection.
The NEO iDSD is internally a fully balanced design, so its XLR outputs are recommended.
The writing/symbols on the OLED display will flip around when placed vertically so that everything can still be read easily.
Not at all. For our X-series products, we designed a highly sophisticated Bluetooth engine that provides remarkably good audio performance, unlike regular Bluetooth receivers. The NEO iDSD features this circuit and also supports highly regarded LDAC codec. While others use the last generation chip, the CSR8675, which is around 6 years old, we use the latest QCC5100 Bluetooth chip. We are one of the first companies to use it. It supports the latest aptx Adaptive, LDAC and HWA/LHDC, together with aptx/aptx HD/aptx LL, AAC etc.
Bluetooth icon flashes. To pair it with your device of choice, find the 'iFi Hi-Res Audio' on the list in To enter pairing mode, press and hold the button (for three seconds) until the your phone/tablet and choose it.
Although the NEO iDSD's volume control is analogue, the volume attenuation itself is done inside of a precision resistor ladder embedded inside of an analogue chip, which is then controlled digitally. So no, the NEO iDSD doesn't incorporate an analogue potentiometer.
IEMatch (when selected) is automatically set depending on whether a 4.4mm or 6.3mm headphone is plugged in. ie. Automatic selection between balanced output and SE output.
IEMatch (when selected) is automatically set depending on whether a 4.4mm or 6.3mm headphone is plugged in. ie. Automatic selection between balanced output and SE output
Currently, our streaming devices can use Airplay, which may not be compatible with certain platforms like Amazon and YouTube, which predominantly use Chromecast technology. For those wanting to use Amazon or Youtube streaming, 'TuneBlade' can be used for PC, while 'AirMusic' is an option for Android users.
Local Network Connection: White LED 'Network Connection Green LED' No Connection Wi-Fi Connection (Network): Green LED 'Speed < 10mbps White LED' Speed > 10mbps Wi-Fi Connection (No Network): Green LED 'Speed < 10mbps Yellow LED' Speed > 10mbps Red LED ' No Connection
The I2S via HDMI connector pinout is: 1) Data '2) Gnd 3) Data + 4) Bck + 5) Gnd 6) Bck 7) Wck' 8) Gnd 9) Wck + 10) Mck + 11) Gnd 12) Mck 13,14,15,16) DSD Enable; L=PCM, H=DSD 17) Gnd 18) 5V Power Enable Output; L=O, H=On 19) N/C 20,21,22,23) Gnd
Neo Stream optical ethernet input is compatible with SC optical connectors using PC,??SPC or UPC connecting surface. Neo Stream is NOT compatible with SC-ATP (typically green coloured cable end) angled optical surface which can permanently damage the Neo Stream and the optical cable.
At this present time we are only producing the cable in 1.8m lengths.
The cable geometry has the live and neutral cables opposite each other and the earth wire placed centrally in the zero field zone.
The Nova cable is rated at 15 Amps up to 125v and 10 Amps up to 250v.
We are working to introduce these cables into the UK market before the end of this year (2021). If you import a cable into the UK please be aware that you need to use a Schuko to UK adapter with earthing.
This configuration results in better immunity to noise, both through shielding the active conductors and through less induction into the earth wire.
Is this OTG cable compatible as a file transferring adapter? Answer: OTG is 'On-The-Go' and is similar to a normal standard USB cable, except an additional pin is connected via a resistor. This resistor tells a phone or smart device, which is normally in 'device mode' pretending to be a USB disk to switch into host mode, where it acts like a computer and can (for example) drive USB connected DAC's.
OTG is exactly a standard USB cable, except an additional pin is connected via a resistor. This resistor tells a phone, which is normally in 'device mode' pretending to be a USB disk to switch into host mode, where it acts like a computer and can (for example) drive USB connected DAC's.
If you cannot connect your Pixel phone via the OTG cable, please go to Settings > OTG Storage > turn setting to On.
This is because Apple cables have a chip inside to release the digital extraction whilst Android is different, its an actual resistor to short out a signal path. Therefore one must use the official Apple cable.
As 'OTG' adapters are also USB C to USB A converters for USB 2.0 only, they could be used on a MacBook, but again, you would need another cable to go into your DAC if not USB A female (for example, iDSD nano BL).
Depending on the country:
Main Outputs: <= 2 Ohm Unbalanced & <= 4 Ohm Balanced 3.5mm Outputs: <= 3 Ohm unbalanced & <= 3.4 Ohm Balanced
It uses DC. Also, filament current is designed for 5670 Standard, which is 0.35A @ 6.3V+/-10%. Do not use tubes of other ratings. Also 5670 has a very unique pinout (RETMA code 8CJ) that is not shared by other tubes. Only 5670/2C51/WE396A tubes that share the RETMA 8CJ pinout, and are electrically equivalent, can be used. Any tube that is pin-compatible and electrically compatible with the 2C51/5670 will work. WE396A are often co-marked 2C51, they will most certainly work. Pin-out in the iFi products are all for 5670, so no adapters needed. Please note, tube rolling or opening the unit will void the warranty.
Every mains powered product is subject to tiny amounts of leakage of electricity from the mains. These can cause very mild electric shocks. It is a sign that the whole system is 'floating', meaning without any earth/ground connection. In this case, it is necessary to add a supplementary earth/ground to the system. You should consider our??GroundHog+ device. The Pro iCAN/Pro iCAN Signature is designed to operate without requiring an earth/ground connection for electrical safety, however for best audio quality and to avoid these mild shocks an earth/ground connection for the system is required.
The Pro iCAN/Pro iCAN Signature is inherently balanced and offers only a single pair of balanced channels, optimised for sound quality. Having multiple independent headphone outputs means essentially having two fully separate amplifiers. Some low-end headphone amplifiers are built like this with four, or even eight separate headphone channels. If using multiple Pro iCAN/Pro iCAN Signature?? is not possible (expense, etc), individual headphone amplifiers per musician are recommended. The intention of the Pro iCAN/Pro iCAN Signature is to offer the recording/mastering engineer (or the audiophile) an ultimate monitor system using headphones while providing an experience akin to large monitor speakers or indeed to drive these speakers (or both).
With a Japan Alps rotary volume potentiometer (and remote control), it is the '6-Way' version with 4 tracks used for balanced volume control. This is the litmus test for any amplifier if it is truly balanced as others that are 2-way are not truly balanced. So yes, the Pro iCAN/Pro iCAN Signature are fully-balanced from beginning to end.
When using the Pro iCAN/Pro iCAN Signature in pre-amplifier mode and go from Tube to Tube+ there is no sound but there are red LEDs flashing. In addition to this is started whilst in Tube+ mode it is find, the solid-state mode is fine too. Answer:??Red means the DC protection has engaged, possibly one tube is bad but the tube determines the DC operating condition. If they're outside the'safe'zone the system will assert DC protection. There are four tube sections (2 channels X 2) and if one is'borderline'it could cause the behaviour (pre-amplifier mode from Tube to Tube+ doesn't switch correctly). If you seem to have these symptoms please open a support ticket.
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The Pro iCAN/Pro iCAN Signature will switch the outputs off and turn the iFi logo red if there is excessive DC detected. There is no way to distinguish asymmetric clipping with music from DC. Indeed, severe clipping actually puts something close to DC into the headphone and is really not a good idea. If the Pro iCAN/Pro iCAN Signature 'red-lights', simply back off the volume until it recovers. Unless the level is lowered, the Pro iCAN/Pro iCAN Signature will continue to clip and thus keep the protection circuitry engaged. If there is severe over-current the Pro iCAN/Pro iCAN Signature may shut down completely, to prevent a meltdown. As there is absolutely no conventional protection circuitry, current limiting etc. otherwise in the actual signal path. So over-current shutdown and output DC detection are the only safeguards for headphones to protect from potentially 32V DC which will fry them in short order. When adding bass boost, obviously more power is required. The full bass boost from the Pro iCAN/Pro iCAN Signature at the given frequency will require 16 times the power of the mid-range. So this feature must be used sensibly. A lot of modern music has most of the energy around 40Hz-80Hz. Simply slapping a huge boost on this when the amplifier is already running near the max is going to trigger protection circuitry. As for Tube+, again, in Tube+ mode has much higher levels of harmonic distortion and will clip a little earlier, as the whole circuit is re-configured (it is not some effect gimmick) to reduce negative feedback to only a few dB. So in this case, if you have the volume very high, the protection circuitry will engage. The Pro iCAN/Pro iCAN Signature is best considered as a racing car. It does not have the most fluffy consumer system safeties. So it will allow you to combine settings (e.g. high gain, high volume setting, high bass boost added to a high 3D setting) that will cause a crash. Except in the Pro iCAN/Pro iCAN Signature, the protection circuitry will shut it off at the edge of crashing to avoid damage to either the amplifier or headphone. So take the foot off the gas and it will start up again. Remember, with great power comes great responsibility, so use the functions and gain/volume settings responsibly.
We are going to provide an example by using the AKG K1000. This is extremely difficult to drive without applying sound corrections. If using XBass at 40Hz setting, that means 40Hz (where for example E-Bass and Piano notes are strongest) 12dB (or 16 times the power) is needed for the 40Hz and below frequency band compared to the midrange around 1kHz and compared to the 40Hz and below frequency band without X-Bass. The 3D matrix also consumes extra power at low frequencies. So combining maximum XBass & 3D settings means the overall loudness must be reduced to ensure the amplifier is not overloaded. If you see a red light it means exactly that. The Pro iCAN/Pro iCAN Signature?? amplifier is being overloaded and the solution for this is to use less extreme settings for 3D and XBass.
If using the balanced outputs, can the non-balanced be used to drive a sub-woofer? Answer: Yes, however, line-level feed to subs just does not work (not even in 5.1 HT systems with proper bass management). Any subs should be driven from the speaker signal, never from a line-level signal if correct integration with the main speakers is required.
Any standard programmable/learning remote (e.g. Logitech Harmony) should work.
Because of the way the 3D sound matrix works any balance control would have to be implemented after this processing. As part of this processing is implemented directly at the headphone output this is in effect not possible. For some, the only solution is not to employ the 3D sound matrix.
There are several issues. 1) With headphones connected the sound quality of the pre-amplifier output may be reduced (depends on the precise headphone(s). 2) If an amplifier is connected to the line out and switched off (to cut speaker output when using headphones) the sound quality of the headphone output may be reduced (depends on the precise amplifier used). 3) There is no independent control of volume or selection of headphone/line out.
Bit-perfect+ is only available for sample rates of 96kHz or lower. If the PC sent a different sample rate (e.g. 192kHz) there is no Bit-perfect+ option.
It is to power external devices, mainly the Pro series using only 1 power supply. However, the DC loop is not 15V/4A. It is the difference what the powered unit consumes and what the external power supply unit can supply. For example, idle the Pro iCAN/Pro iCAN Signature consumes 22W, so only 38W of the 60W the iPower Plus/iPower Elite (15v/4A) can deliver are available to the looped device, less if driving difficult headphones with high power.
It has a self-resetting poly-fuse. Important to understand, the Pro range converts incoming power down to very low noise and highly regulated 5.5V DC. Then this DC is used to create all the new voltages needed for the system. Again, each voltage is individually regulated noise filtered. If there are over-voltages etc it may trip the fuse, which would reset itself after a while.
The XLR portion is wired as L/R connections in accordance with the system originally introduced and promoted by Headroom USA for balanced connections. Pin 1 'Ground Pin 2' respective channel Positive (+) Pin 3 'respective channel Positive (-) This is the standard XLR pinout. This very different from the wiring of the 6.3mm jack portion, which uses one connector for stereo, the left one with the negative polarity output from the amplifier and right one with the positive polarity (normally that connector should be used for critical listening). This 6.3mm connection allows standard headphones to be connected using a single 6.3mm jack connector (to either 6.3mm jack) while with the correct wiring two 6.3mm jack connections form a balanced connection. Hence'Single Ended Compatible ' Balanced.
The Pro iCAN/Pro iCAN Signature can comfortably handle way more than 3V, certainly, +26dBu (15V) is fine.
With switched resistors (if using mechanical switches and so on) the adjustment range is very limited and steps are large. If using electronic switching we prefer a good quality traditional volume control.
We have tried a file sent by one of our customers in Muzo, it does play weird. We've sent it to the Wi-Fi module vendor to check. When playing with Muzo, the audio is decoded in the device side, the device's decoder is not fast enough to decode the file.?? The Wi-Fi module's performance is limited.
For iDSD Pro and the Pro iDSD Signature, use a hardline network for playing High-Res. Wifi is not that reliable in this case. The iDSD Pro and Pro iDSD Signature support UPNP/DLNA. UPNP support means a wide range of apps can be used for server and control, including those provided by competing brands ' which to mention a few: AudioNet Remote Control Point: http://en.audionet.de/apps/rcp/ Linn Open Source Apps: http://oss.linn.co.uk/ Payware (including DSD) also exists. dBPoweramp Asset UPNP: http://www.dbpoweramp.com/asset-upnp-dlna.htm This can transcode DSD to DoP and send onto iDSD Pro/Pro iDSD Signature In terms of UPNP servers control surfaces there is an almost infinite profusion. People should just use what they are familiar with and like. For Apple, use either the Mac/iOS versions (where available) of the above or just use Airplay. We have confirmed playing 192k using Audionet RCP & Asset UPNP works just fine.
In the Pro iDSD/Pro iDSD Signature we are running 4 actual DACs in parallel with interleaving, meaning the net result is closer to 8 bit via multi-bit. That means the top 36dB (6bit) / 48dB (8 bit) are handled by a multi-bit architecture. As delta-sigma modulators struggle at high levels, we are removing the main disadvantage of DS, and we are avoiding the need for high order modulators. Now multi-bit systems have the opposite problem of delta-sigma, meaning the are not very good at very low levels, and multi-bit struggles most at very low levels (around zero crossing). By splitting low level signals off into a low order delta-sigma modulator, you get the best of both worlds. So the top 6-8 bit in iFi gear are processed as multi-bit, including the option to apply no digital filter. The lower bits are processed as delta-sigma over-sampled to what is in effect DSD256 (11.2/12.3MHz) again, it is possible to not apply any digital filtering. This whole topic was covered here: https://www.audiostream.com/content/qa-thorsten-loesch-amrifi Around half way down the page, search for: 'The DAC chip we use in the iDSD nano offers a rather unusual way to handle things.'
The Pro iDSD/Pro iDSD Signature can be affected by packet errors the same way any USB device can. It is not, however, affected by noise on the USB power (it does not use USB power) or ground/Earth loops via USB or other inputs (because it uses galvanic isolation).
As MQA controls the time and supersonic frequency-domain behaviour of the system all the way to the analogue output, there are no options for filtering or up-sampling available, as they would fundamentally run counter to the aim of MQA to deliver a reliable and certified 'studio master' result to the analogue outputs.
When playing music through the device's USB input, the laptop freezes up after 1/2 hour or so of play and requires a hard boot to restart. By process of elimination, if the laptop Wi-Fi is turned off. no freezing up occurs. Answer: It's possible. Drivers can 'clash'. We would suggest looking for the latest drivers for the OS and allow Windows updates to run. This often solves such issues.
The interval between music changes, it has a lot to do with the speed of the network, the router is connected to the mobile phone, and the router is connected to the Pro iDSD/Pro iDSD Signature, so there will be a delay. We tested the connection network through wired, and the maximum test interval was 1.5 seconds.
There is no 'headphone amplifier' in the Pro iDSD / Pro iDSD Signature. Just a gain stage/line driver that is sufficiently powerful to drive many headphones directly, however, doing so is clearly a compromise over using a dedicated headphone amplifier like our Pro iCAN/Pro iCAN Signature. It is adequate for anyone occasionally using headphones while mostly using a speaker based system. However, unlike the Pro iCAN/Pro iCAN Signature, it is not optimized for headphone drivers. It cannot drive as wide a range of headphones as the Pro iCAN/Pro iCAN Signature and it lacks the sonic controls offered by the Pro iCAN/Pro iCAN Signature. It should be seen as a state of the art DAC, which may drive headphones, and not as headphone amplifiers that accidentally had some 'normal DAC' fitted.
https://www.dbpoweramp.com/codec-central-dsd-dff-dsf-sacd.htm
The whole Wi-Fi subsystem has its own firmware. It outputs audio via I2S to XMOS. Other than turning the thing on/off WiFi module operates 'standalone' on its own software. If stutter is present with Wi-Fi but not LAN, the issue is 99% certainty poor Wi-Fi. Stutter on WiFi but non on wired simply means WiFi spectrum is congested and the signal poor. If you expect generic Wi-Fi to carry full fat audio (not MP3 or the like) without stutter, you need a clean 2.4GHz spectrum. For serious audio, hardwired please, Wi-Fi signal quality is not reliable. But that has nothing to do with iDSD Pro. If it works on a hardwired network, but not Wi-Fi, the problem is the Wi-Fi signal. An easy way to see if you have a good chance for major Wi-Fi problems is to open your phone's Wi-Fi screen and count the number of networks shown. If more than one and your router has not been set to use'non-default'Wi-Fi channels (you would know if you changed those settings, if you didn't they are 99% default), you have problems. Almost all routers default to the same channel. We believe in 2.4G and the highest rate at 'n' ' ONE router/device pair in an empty spectrum has enough bandwidth to stream 192/24. Even having anyone surfing the web on the same Wi-Fi may be too much. Neighbours networks on the same channel steal bandwidth, as the spectrum congests.
Some SSD/USB memory sticks will work powered from the Pro iDSD/Pro iDSD Signature 'A' (host) interface, however, this cannot be guaranteed, as some drives require as much as 1A. If in doubt, add iDefencer+ and iPower to provide power to the drive USB.
Does the fixed output bypass the analog pot completely, and does the tube get used on the DAC's output stage, or is it for the amp stage only? Answer: Yes on both.
In solid state mode, the tubes are turned off (heater off) and disconnected from the circuit. In tube mode the tubes are active and the solid-state device is disconnected from the circuit.
No, not at this time only 2.4 GHz.
No, oveninzing a crystal merely reduces long-term thermal drift (over periods of minutes to hours) which is inconsequential to audio playback. We employ a high quality miniaturized 10MHz discrete crystal (not canned oscillator) as a timebase for our GMT clock system. The GMT clock system allows the clock frequency to be set to appx. 0.01ppm (parts per million) or better than 0.5Hz accuracy compared to the nominal 45/49MHz audio clock frequencies.
No, in the Pro iDSD/Pro iDSD Signature we operate it very differently. We use a suitable isolation barrier between the digital (noisy) section with XMOS, WiFi/Networking and the Chrysopeia and the DAC's Reclocking / Clock. In essence, all noisy digital processing is confined to a blocked off'island' on its own board and isolated from all audio circuitry. This parallels the way in which for example the JVC K2 system is implemented for playback and the isolation of the DAC from digital noise found in the Legendary Marantz CD/DA-12 system featuring the TDA1541. All this is essentially trickle-down from the DP-777. The nano iGalvanic3.0 isolates the USB connection instead, so it is a less complete solution, as it cannot isolate noise inside the DAC itself.
Yes, however, as the iDSD Pro is galvanically isolated on all digital inputs this is less of an issue WRT Earth Loops.
Green?? ?? = Warming Up, White?? ?? = Solid State, Orange?? = Tube Mode Red?? ?? ?? ??= Protection Mode
The Pro iDSD Signature can process MQA over WiFI using the MConnect HD App and playing from Tidal. The Linkplay module receives the wifi signal and sends undecoded MQA to the Xmos chip, where it's decoded.
OFS is a valid state of MQA indication for when a device receives an MQA signal already decoded by an upstream piece of hardware or software. As signals decoded upstream cannot be authenticated in the same manner as the original un-decoded bitstream, we use 'OFS' which stand for 'Original Frequency Spectrum' to show this is still an authenticated MQA stream, and the end point device is rendering correctly, based on the Original Frequency Spectrum. If your audio source is an upstream application with an MQA decoder, such as Tidal or Roon, the application may have a switch to disable the MQA decoder and passthrough the bitstream signal to your endpoint device. Decoding in hardware or software, splitting the process or performing both in the same device, will not give any audible difference, as the decoding and rendering information is the same, based on the bitstream content. In essence, the system is working correctly and MQA content is being received and rendered correctly if you can see 'OFS' indication.
Could a 6.3mm to 4-pin female adaptor from an un-balanced amplifier be plugged into the rear balanced connection? Answer: Yes, that setup will work.
The wiring is based on the HDMI standard, and any 100% compliant cable will work. There is no 'special wiring' with our provided HMDI, but it's a good quality one not to be discarded! However, many commercial cables omit connections used for the iESL link, and those often do not use sufficiently high-quality wire for these connections.
The LED is green, there was a typo in the manual. Please also remember that there are LED's that are orange to indicate the unit is ready, so its possible that this may shine through the'green' LED section.
On 16 Ohm setting output = input, on 64 Ohm setting output = input/2. The 24/94 Ohm settings add some impedance to tailor the sound and may reduce the output voltage slightly, depending on headphones over 16/64 Ohm settings.
We recommend an output of at least 10V into 16 Ohm or 20V into > 16 Ohm to drive the iESL.
What specifications should be met by an amplifier to pair it with the iESL Pro to properly drive electrostatic headphones? Answer: Recommended is 25W/4Ohm to 100W/8Ohm power. This means, for example, an amplifier rated as 7W/16Ohm, like a 300B SE amplifier with a 16 Ohm loudspeaker output will be fine. Equally, a solid-state amplifier with a 100W/8Ohm rating will be fine and pretty much anything in-between. Amplifiers with much higher power may cause overloading and damage the headphones if the volume is raised too high.
A lot louder'likely means at a given volume control setting, which is meaningless. For the whole chain the key elements are: Headphone/Speaker amplifier' Output voltage is controlled by volume control. Energiser 'Output voltage is controlled by the driving Amplifier AND the step-up ratio. We deliberately use a low step-up ratio (1:16/1:32), as with increasing step-up ratio transformer quality becomes more compromised with increased distortion and limited frequency response. The original Stax energiser adapters are rated to produce full output from a 5 Watt / 8 Ohm Amplifier (or 6V), suggesting a step-up ratio of 1:48, so naturally, they will be much louder at the same volume setting on the HP Amplifier. Hence swapping between the iESL and any other'Energiser' is only valid if the volume control is corrected for the difference in step-up ratio. To consider this as car alternative, Stax and iFi Energisers are like gearboxes with different gearing ratio, so even if the same engine the gas pedal will need different degrees of being depressed for the same speed.
Standard 4-Pin XLR pinout as per AKG K-1000 https://cdn.head-fi.org/a/359627.jpg
The foam we use is quite unique. At low frequencies (where the reflex system operates) it has nearly no effect. So bass is not damped, but by the time we get into the mid-range and frequencies, the enclosure has parasitic resonances from the shape. It absorbs most sound. With a thin and light paper cone, any sound reflected from the back of the enclosure would bounce through.
The tweeter is horn-loaded, not the bass. The bass loading is a variant on the 'double chamber reflex' system. The earliest note of this comes from P.G.A.H. Vogt, but it has been somewhat popularised by Augspurger (USA) and by Kondo & Nagaoka (Japan) especially for smaller wideband drivers.
4 Ohm.
When all the 3 LEDs light up and you've excluded the tubes by using the supplied extra tube. This means the front PCB gets no signal or the MCU is not starting up. Please open a support ticket with us if this is the issue.
Sonically the greater effect is always the driver tube, compared to output tubes. NOS EL84 should remove the four jumpers. They should also be fully tested, including at rated voltages, to be sure they work 100%. Given how desirable a tube the EL84 is (use with many different home and guitar amplifiers) it is EXTREMELY UNLIKELY??to get good quality NOS tubes. There are generally two possibilities for western NOS versions of popular tubes ' they are rejects of multiple selection rounds, and are in effect the remaining rejects. In which case, often the performance is not acceptable, or they are outright fakes (which should be avoided at all cost). It is not possible to exclude 100% the possibility of buying good NOS EL84, but it is very small. On the other hand the ECF82 is primarily a TV tube, and rarely used in audio (though it is very similar to 7199 which is a widely used audio tube and hence again extinct in NOS). So large NOS stocks of good tubes remain. We would suggest to leave the EL84X factory supplied in place, and only use NOS options for the ECF82. There multiple equivalent tubes, any tube equivalent to ECF80 generally also works in place of the ECF82 in our circuit, but has slightly different operating conditions, and hence sound quality. Please note, any tube rolling will void the warranty.
The SPDIF iPurifier will follow what the source does, as it offers a bit-perfect pass-through. So if the source drops the signal there will be no output.
With regards to the Bluetooth can you try the following; 1) Manually delete the connection from your smart device. 2) Turn off the Retro and smart device. 3) Restart the Retro first and let it warm up. 4) Start the smart phone. 5) Move the toggle to Bluetooth. 6) Keep the input selected as Bluetooth. 7) Hit the Bluetooth button on the back of the Retro. 8) Re-establish the connection. 9) Play audio. Does it work? If not please open a support ticket. Tip: Phones differ greatly in Bluetooth operation. Some (e.g. Sony XPERIA Z series) have great range (>10m without walls on Retro). Some others (e.g. many/all iPhones) have very poor Bluetooth range and need the phone to be within <2m of a Retro and not to be handled (this is the Bluetooth version of antennae gate and was never fixed), especially if the Phone's wifi is on. We would suggest to benchmark the Bluetooth connection using a Sony Xperia Z series phone, it should have stable connection within the same room (BT antennae fitted and vertical) and > 5m distance. If so, the problem is not the Bluetooth inside the Retro. Also observe that the Retro has no earth connection, a supplementary earth connection (Groundhog) may improve Bluetooth performance.
Our own experiences with 'burn-in' (that is a material parameter change in electronic components over the early hours of being operated in a circuit after being manufactured, or inactive for a substantial time) caused us to investigate. Here some samples of the literature we found: Others exist and the above list is neither complete nor exhaustive. We hope this has provided some insight.
The Stereo 50 was designed ground-up without earth connection. There is no earth connection in the IEC Socket, it is a C18 Standard type, that readily accepts power cables without earth connecton. If you wish to add an earth / ground you can use the Groundhog from iFi.
The circuit of the Retro is auto-adjusting the bias, no manual adjustment needed. Just replace the tubes (not even matched pairs are needed, though it measures better if they are used and we match output tubes quite closely as quads) if they tire out. They should last between 5,000 and 10,000 hours, at six hours per day this is around 3 years (based on 7,000 hours lifespan), at 12 hours halve that and so on.
The PCB has a jumper for EL84x tubes. What does the 'X' stand for? Answer:??We use a tube that is a modern equivalent of the EL84. It has a slightly different pin-out from the standard EL84. In most cases you can just plug them in, but with the correct external circuit, the EL84X can give more power than a standard EL84. For modern tubes that are all based on the same design (Russian, Chinese and Czech) and quite a few older EL84 this is no issue, but for example the Mullard EL84 (and some others too) have an internal connection on a pin that is normally 'NC' (no connection) that could lead to fireworks. Some commercial EL84 amplifiers actually use this pin as a wiring point (in line with the EL84X pin-out, incidentally or intentionally), but they cause trouble with some old tubes. We like to avoid that. Pulling the jumper makes the Stereo 50 compatible with all original old stock EL84 (in case you can find any), with the jumper in place is works with the EL84X and delivers deliver more power.
The Retro is a Class II or double insulated electrical appliance. It has been designed in such a way that it does not require a safety connection to electrical earth (ground). This was done to avoid any chance of earth loops if (for example) Satellite or Cable TV systems are connected. So it was intended that there is no Earth connection nor is one required. The presence of high voltages inside the case is inconsequential, as without an earth connection this voltage can only ever offer one pole, even with a short to the case, there is no second pole to complete an electrical circuit. From a safety viewpoint the Retro is hence safer than Valve Amplifiers WITH an Earth Connection, as in this case a second pole exists. Testing Procedure: We use an industry-standard insulation tester to test the Retro according to the required standards for double insulated equipment. This is according to EN 60065 and we also test leakage current according EN 60065 for Class II equipment. For completeness, we additionally also test according to Medical Equipment directives, even though they are not required to pass.
6.3mm:??< 6 Ohm 3.5mm:??< 3 Ohm
No AVRCP confirmed.
The output transformers will produce a small sound, especially if not connecting load. Even output transistors in solid state amp's tend to'sing'along with the signal, though the mechanism is different. Capacitors (e.g. in Speaker Crossovers) often have a similar issue. All electronic components are subject to microphonics and with signal applied reverse microphonics. If the sound output is material depends on many factors. The job of the transformer is to transfer power into a load. Without load the power has no-where to go. This causes electrical and mechanical resonances in the transformer to be'fed'with spare power which would normally be damped by the load. This will make microphonics worse and further can create very high voltages (resonance' no load), even high enough to create a miniature lighting strike inside the transformer (which usually burns away the isolation and renders the transformer defective). Modern isolation such as we use at iFi should handle this, but better not walk on the wild side. So a few rules here: 1)??Never operate ANY tube output amp with output transformers without load. 2) At high playback levels some 'singing along' noise from power amplifiers is expected.
Short answer:??The typical 'measured' max output power is 15-17W. This is measured in a lab read: simulated, with a sine wave into a resistor load. This is not reflective of what the real world loudness is, not the actual load. Longer answer:??It is not always advisable to measure valves and solid-state using the same 'in lab' approach. For example, Solid-State doubles the power each time impedance drops. With valves, power stays the same when impedance drops. They just are not the same. Therefore, we have done our utmost to 'equalise' the measurements by using an actual 'music signal + speaker load' so the real world measured power is the same loudness (SPL level) for solid-state as it is for valves. Most important of all this approach means a customer will find in reality, with the same music signal and on the same speaker, the iFi Stereo's 25W+25W is the same loudness as a 25W+25W Solid-State amplifier. Not louder nor quieter. The more lengthy explanation is to be found here: https://downloads.ifi-audio.com/audio_blog/retro-stereo-50-there-are-watts-and-there-are-watts/
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Yes, the Retro is a Class A headphone amplifier too, therefore no harm if no speakers are attached.
Transformer coupled.
The power supply in the Stereo 50 is a regulated SMPS, designed for 85V ' 265V operation and the design has been verified against these voltages for normal operation. Example:??The mains supply in Japan is at the lower edge of this range, with 100V nominal and 90V minimum, however even for Japanese mains there is a safety margin.
Firstly the power supply in the Stereo 50 is a regulated SMPS, designed for 85V ' 265V operation and the design has been verified against these voltages for normal operation. Example:??If we use Japan as an example country.??Even given the mains supply's regulation, the low voltage in Japan will lead to increased current in the power supply and so to increased heat. The power-supply includes over-temperature protection which will cause shut-down in case it gets hot to endanger the safe operation. We have carried out long term tests at 90V, without experiencing any problems. If the unit is operates in a fairly hot room (summer, no aircon) and it's ventilation is impeded it may enter thermal protection, which is self-resetting, so the unit may drop out and come back after a short time, work for a short and then drop out again.
Unfortunately, there is not and will not be a line out. On this basis alone, the Stereo 50 is unsuitable for Magnepan's. Magnepan's A.K.A 'Maggies' are 86dB/2.83V with 4 Ohm impedance, so they come in at around 83dB/1W/1m. To play at 105dB peak (THK recommendation and in line with realistic playback of real uncompressed dynamic range classical music) a pair in a normal sized room and with around 3m listening distance will need in the region of 400W peak power available. So the Stereo 50 (25 W / 4 Ohm) will cut it.
Yes. Sub-woofer with high gain connected to the speakers. We always recommend that a sub-woofer is connected in 'high pass' mode that is, where it is connected to the speakers or at the outputs of the amplifier. Depending upon the sub-woofer, this usually requires a cable that is co-axial at one end (to the sub-woofer) and spade at the other end (which is connected to the speaker outputs at the amplifier or speaker). This is a good description: https://www.cnet.com/how-to/how-to-hook-up-a-subwoofer-to-a-stereo-system/
This means that the tube has a fault and the tube protection system is flagging that its the one triggering the machine to reset mode. Please see this image, its likely lightning in a tube. You need to remove it and replace it with the spare in your accessories.
The Retro LS-3.5 speakers are what we recommend. Not just because we designed them from the ground-up but their 90dB/2.83V sensitivity makes them a perfect match for a tube/valve amplifier such as the Stereo 50. The 90dB/2.83V is for approximate 30m^2 / 300ft^2 rooms. For larger rooms increase the sensitivity requirement by 3dB for each doubling of the floor area, so for 120m sqm / 1300sq ft, the minimum sensitivity should be 96dB/2.83V. This allows around 100dB undistorted peaks. The common peak-to-average ratio of music (classical, tutti/crescendo) is around 14dB, so 100dB peaks allows an 86dB average, which is the normal SPL in a mid-hall seat in many concert halls. Obviously this formula is an approximation and if one likes to listen to music very loud, the efficiency required may be even higher. Otherwise we recommend a??minimum of 90dB/2.83V sensitivity??(real 'independent 3rd party tests preferred) and a nominal impedance of 4' 16 Ohm with a minimum impedance no lower than 3 Ohm. The 90dB/2.83V Sensitivity is the key number to bear in mind when selecting a speaker to match with the Stereo 50. Question: What about mainstream bookshelf speakers? Answer: Most mainstream bookshelf speakers were designed for 100W> amplifiers. Hence despite their diminutive size, they are still not suitable for the Stereo 50. Typically, such speakers range in actual sensitivity from 82dB/2.83V to 85dB/2.83V. This makes them a less than ideal match with a tube/valve amplifier such as the Stereo 50.
Is there any benefit to using more than one of these at a time, stringing them in a series? Will it eliminate even more jitter? Answer: Not really, however, on the other hand, if using very long runs of SPDIF cables, putting one, say, every 10m or so would do good.
As the buffer rejects jitter, it's length in the time-domain varies according to incoming jitter and sample rate. The actual length at 44.1kHz, with no jitter input, will be around 10mS. Short enough to avoid lip-sync issues. Around 10mS at 48kHz, it will be slightly less. The higher the sample rate the shorter the delay In the AMR DP-777, we were very careful to balance off the rejection of'wander', essentially jitter at sub-Hz frequencies, (which calls for a very large buffer) and lip-sync issues (which call for a short buffer). The technology in the SPDIF iPurifier derives from the AMR DP-777 and thus, just like the DP-777, uses a buffer short enough to avoid lip-sync issues and yet long enough to allow even wander to be contained well.
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Answer: The SPDIF iPurifier is a totally different design. It shared with other iPurifiers only the aluminum case and the concept, purify something very dirty, USB, DC, SPDIF or whatever it may be. The best way to look at the SPDIF iPurifier is to see it as a modern-day take on the late 1990's'Digital Lense'concept. Incoming SPDIF is isolated (unless optical, that is isolated any which way), then the waveform is restored using a solid-state implementation of the Valve High Definition Digital Input first seen on the AMR DP-777. Then the restored digital signal is sent into a memory buffer. The same precision adjustable 300 femtosecond jitter clock found in the DP-777 is used to match the incoming clock average and clock out the data from the buffer' minus any jitter. What it means is that directly plugged into the input of your DAC you get a SPDIF signal with perfect waveform and as low jitter as a high crystal clock can provide. It won't get better than that, of course, your DAC remains the limit. Different DAC receiver chips have different inherent jitter, from the common Cirrus Logic Chips with 200pS jitter to the AKM and TI with. To get those 50/200pS jitters you must feed them a zero jitter signal, any added jitter rides through. Of course, the SPDIF iPurifier removes all source jitter and adds very, very little of its own.
Any impact from good digital quality cables should be lowered significantly. However, if cable quality is too low then there may be issues at higher sample rates, so reasonable quality cables should be used.
The output of the SPDIF iPurifier is direct-coupled for the precise reasons the customer states. If no isolation is needed this is best. Further, however, in any competent design one cannot simply bypass or exchange pulse transformers and expect correct or improved performance. In a fully optimised design (such as we employ) additional complex conjugate pole and/or zero circuits are added to maximise transformer bandwidth and return loss across the widest possible frequency range and must critically tuned to the actual transformer used. As a further note, pulse transformers that perform best for SPDIF generally use bifilar (aka. high capacitance) construction on high-frequency ferrite cores of small size. Using amorphous iron or permalloy cores (or other 'exotic' metals) in SPDIF transformers (as some manufacturers promote) or a shield between windings (as some other manufacturers do) is not conductive to good performance but makes for good marketing spiel that allows high prices to be charged for very limited performance and those transformers are best avoided. In AMR & iFi products with isolated SPDIF inputs, we use a specific wideband pulse transformer fully tuned with complex conjugate pole and/or zero circuits for a very wide bandwidth and maximum return loss for the widest bandwidth while providing isolation on the input. This is followed by a linear mode, zero-feedback active circuit (not logic circuitry) using either tubes (AMD DP-777) or Fets (iFi) with a designed-in 'clipper' circuit that regenerates a near-perfect Logic waveform even from badly degraded SPDIF signals. It is this regenerated signal that is fed to the rest of the circuitry. So the presence of galvanic isolation is not an issue with AMR or iFi products. More on this fundamental technology here: https://amr-audio.co.uk/tech-notes/#1539092351973-29777ab8-bf2c
Laserdisc RF demodulator output is not SPDIF compatible and so will not work with SPDIF iPurifier or indeed any SPDIF input.
SPDIF iPurifier and the noises without signal with certain DAC's could well be an issue but if used in a CD player, then can rule in/out so please try a CD and report it to us via a support ticket. Past that the most likely explanation is what these systems have in common is: Example:'Chrome Cast Audio with TIDAL in Chrome as source.'As opposed'it happens with Brand X DAC and any source' We believe that the SPDIF iPurifier merely follows the source (CCA from our example) which in turn receives garbage from the (windows audio) driver and outputs it. It could be that the garbage is created by TIDAL itself, possibly switching sample rates/formats. The whole chain via Tidal Chrome Plugin, Google Chrome Browser, Windows Audio Driver, Windows Network Driver, Chromecast Audio, SPDIF iPurifier and DAC is very complex and has plenty of areas where the signal can be mucked-up. Our suggestion is to try different playback software and a playback path. Note: Our understanding is that some DACs correctly mute the garbage created when connected directly, but not via SPDIF iPurifier, others mute correctly no matter what. Even between the same brand and their different models. But this is best confirmed directly with the DAC manufacturer. If no luck with the above please open a support ticket.
The iPurifier SPDIF does not perform any conversion. However, without input it will normally stay locked on the last sample rate received, if there was no input at all ever it will 'hunt' for a signal to lock on. If there are noises during normal playback with a constant sample rate and during the same track, the problem may be external transients. Due to the galvanic isolation, the output from the SPDIF iPurifier floats without reference to earth. The DAC or one item in the rest of the system the DAC is attached to should have an earth connection. If this does not exist a supplementary earth strictly for audio purposes should be added ' consider the GroundHog+. If the noises only happen during sample rate changes or track changes them the problem is that the attached DAC does not mute correctly and the short time in which the SPDIF iPurifier re-locks and re-buffers the SPDIF signal suffice to cause noise. In this case we can try to evaluate the interactions further, but we must know the details of the DAC. Open a ticket and provide the information.
The SPDIF Purifier is different from virtually all other 're-clockers' on the market. SPDIF iPurifier kills jitter dead without altering data, bit-perfect (so, for example, HDCD, DTS, and MQA will pass unmolested). In that sense the SPDIF iPurifier is a true re-clocker, it re-clocks the signal without altering it and it is as far as we know the sole and only product in the market to do this.??Of the ones in the market sold as 'SPDIF re-clockers' are in effect an up-sampler that takes any incoming audio and convert to a fixed sample rate. As they perform asynchronous sample rate conversion they shift the source jitter into the data domain and embed the source jitter irremovable into the new audio date created, they are not bit-perfect but instead break any non-PCM signal on SPDIF and strip HDCD/MQA information from the signal they pass. Additionally, they process the signal with added digital filtering which effectively overrides that of the DAC. If this effect and processing are liked, then placing the SPDIF iPurifier before the up-sampler can help to dramatically reduce the negative sonics of asynchronous sample rate conversion, as the system is fed an essentially jitter-free signal to start with but we prefer bit-perfect!
Unless the SPDIF cable is much longer than 1 metre it is doubtful there will be an advantage that is unless there is a ground-loop in the system, as the iDSD micro SPDIF input is not isolated (as it is mainly intended for portable/desktop use). In this case, the galvanic isolation offered by the SPDIF iPurifier would provide improvements. It should be noted that the spacing of connections on the iDSD micro does not allow the SPDIF iPurifier to be plugged in directly while having the line outs in use, so an adapter cable would have to be added.
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The SPDIF iPurifier is rated to tolerate normal variations of USB voltage, that is it will work reliably from 4.5V to 5.5V. The absolute maximum beyond which damage may be sustained is 6.5V DC.
The ZEN DAC V2 is a fully balanced desktop DAC/pre-amp product with 16 core processing enabling full MQA decoding, and that goes up to 384kHz PCM and DSD256.
Yes, the SPDIF iPurifier is agnostic as to what audio format is packed within the SPDIF stream, it may be DSD as DoP, DTS, Dolby digital or conventional PCM (or indeed any other 'payload'). It buffers and reclocks the SPDIF stream as received and re-transmits it without altering data and so works for any 'payload' embedded in the SPDIF stream.
The SPDIF iPurifier2 is a totally different design. It shared with other iPurifiers only the aluminium case and the concept, purify something very dirty, USB, DC, SPDIF. The best way to look at the SPDIF iPurifier is to see it as a modern day take on the late 1990's'Digital Lense'concept. Incoming SPDIF is isolated (unless optical, that is isolated any which way), then the waveform is restored using a solid state implementation of the Valve High Definition Digital Input first seen on the AMR DP-777. Then the restored digital signal is send into a memory buffer. The same precision adjustable 300 femto second jitter clock found in the DP-777 is used to match the incoming clock average and clock out the data from the buffer' minus any jitter. What it means is that directly plugged into the input of your DAC you get a SPDIF signal with perfect waveform and as low jitter as a high crystal clock can provide. It won't get better than that, of course your DAC remains the limit. Different DAC receiver chips have different inherent jitter, from the common Cirrus Logic Chips with 200pS jitter to the AKM and TI with To get those 50'200pS jitter you must feed them a zero jitter signal, any added jitter rides through. Of course, the SPDIF iPurifier removes all source jitter and adds very, very little of its own.
The SupaNova power cable was designed to be fully balanced, low loss power cable incorporating Active Noise Cancellation (ANC) and surge protection.
The Nova power cable was designed to be fully balanced, low loss power cable.
The SupaNova cable is rated at 15 Amps up to 125v and 10 Amps up to 250v.
The large barrel-shaped enclosure in the centre of the cable contains the ANC circuitry and shows the polarity and earth indications at one end.
While the xDSD Gryphon is off,?? long press and hold the input button, then press the power button and keep it held down. When the xDSD Gryphon powers on it will display the XMOS version. You can also see which version you have in the GAIA app.
Yes, you can. The 6.3mm/3.5mm front output and RCA rear output will both output audio at the same time. Please be aware that the analogue volume control will change volume on both outputs, so adjust volume carefully to ensure neither output receives too much volume while in use.
The EQs operate as follows: Gaming: Improve ambient and footstep sound in games so players can hear approaching foes better; Movie: Enhance conversation, especially in streaming movies where it is frequently unclear. Due to the fact that each of the aforementioned will enhance various mid-range contents, background noise will also somewhat rise. The SNR of the UNO is 113dB without the aforementioned EQs, and while they are active, it still has a very respectable 109dB. In conclusion, there is nothing to be concerned about and everything is fine with the UNO. When utilizing ultra-sensitive iEMs with the Gaming and Movie EQs, a tiny background noise could be detected, but this is normal. If you want to reduce the noise, try using the IEMatch.
This feature allows selection between digital filter options. Hold the EQ button for 2 seconds ' when you see the flashing LED at the top of the device, additional presses of the EQ button will cycle through digital filter options: When the flashing light disappears, this will confirm the current selection.
Yes, the 2.5mm output is always balanced, regardless of input. Like the iCan Pro, unbalanced signals are converted to balanced.
The 3.5mm connections are TRRS and S-Balanced. However, the 3.5mm input should NOT be used with 3.5mm balanced sources, only iFi S-Balanced. The 2.5mm balanced input expects twice the signal level of the 3.5mm S-Balanced input. Only use one of the two inputs, switch-over etc. is fully automatic.
As the xCan is balanced, if you really want to 'burn-in' the complete circuit then run balanced when possible. If you have an iEMatch 2.5mm this can be left in the 2.5mm port and the balanced input should be used.
The xCan has 2 possible input modes, and 1 output mode: To select the mode you want (input) hold down the centre rotary knob, then let go on the colour that corresponds to the input mode 'it will cycle between blue/green for as long as your holding it down. When in Blue (Bluetooth mode), you then need to pair it with your smart device (phone/tablet), all you do is push and hold in the'gear/BT icon', until the centre rotary knob cycles to blue/red, once paired it will change to the'volume level'. To cycle between 3D/XBass just press the'gear/BT icon'. To power off the device just hold the centre rotary knob until it becomes white then let go.
We've tested a moderately used xCan for operation time on battery, using Fostex T50 on single-ended port playing at elevated listening levels. On Bluetooth with aptX we got up to 12 hours whilst using 3.5mm was up to 18 hours.
We tested an xCan using a pair of Fostex T50 on single-ended and found that on Bluetooth we achieved 12 hours and on 3.5mm line in we reached 18 hours.
It is possible that while on the train (or a form of transport) the phone steps back to 2G networks, this can create EMI. Generally, the frequency is around 220Hz, so not a high pitched shriek, but rather a lower pitch yowl. It is next to impossible to suppress 2G interference, you can also find this with other portable headphone amplifiers. We can speak from experience having tested mainstream products such as the Sony PHA2 on 3G and 4G networks here there is usually no interference at all. However, in Asia 2G is more prevalent and as a result EMI occurs quite commonly. Modern phones generally have settings to make them lock to 3G (and/or 4G) networks only. While this means no phone signal outside 3G coverage, it does make sure no EMI from 2G networks. This avoids the need to use airplane mode, which is the alternative if in situations where 3G network coverage is bad.
For the weird interference noise, please try for a few minutes to keep mobile phones away / turn them off 'this would rule in / out any other device nearby. You can tell as the 3G is disappearing from the phone display. If phones are switching from 3G to 2G networks there is a chatter that is pickup up by headphone amplifiers. It may be the xCan has an issue or it may be the network switching. Newer 4G networks don't have this interference ' it is a legacy thing.
No, it uses the same technology as the xDSD. It has lower noise on the 3.5mm singled ended output than earlier iFi devices. Further question: I seem to hear some white noise of xCan with any IEMs? It depends on the IEM, specifically the sensitivity and also the volume setting. As the xCan has up to 18dB gain with the volume turned up all the way this would produce a lot of noise when using sensitive IEM's. Further question: Wouldn't you say that 2.5mm balanced is noisier than 3.5mm unbalanced? Balanced circuits involve two amplifier circuits instead of one. This allows double the output voltage and up to four times the power of single-ended connections but also produces twice the noise. With a given IEM or headphone, the balanced connection will thus have double the noise.
XBass, XBass II and 3D/3D+ are all analogue domain processing (ASP). We generally prefer to use analogue systems for volume, frequency response corrections and any other processing. XBass/XBass II are frequency response corrections aimed at headphones and/or speakers depending on the product, so they adjust frequency response. On the xCan specifically XBass II offers a frequency response correction in the bass and lower mid-range (presence region of music) that takes a 'typical' open back diffuse field tuned headphone (e.g. most modern European ones) and corrects them broadly for the Olive/Welti Frequency response target, with the option to skip the bass compensation for headphones with an elevated bass response and to skip the mid-range compensation for headphones where the mid-range region is already emphasized (many far eastern ones). 3D/3D+ is not operating by altering frequency response. It is often erroneously attributed as 'crossfeed' and while it is true that it in effect processes the 'difference' signal between left/right channels (like other similar systems) it is not traditional Bauer/Linkwitz crossfeed or Meier enhanced crossfeed (or any of the other published systems based on the same research and targets, be they analogue implementations of DSP). The iFi 3D/3D+ is a separate system to reduce 'in head localisation' and to present a more natural, speaker like soundscape. It is based on very different targets derived in large scale listening tests back in the 1980s in what was then East Germany, far in advance of anything published previously and up to now elsewhere. The results are distinct from and well in advance of 'Crossfeed' of all variants.
Only one input should be used per time, though both 3.5mm and 2.5mm headphone connections are always active and can be used in parallel (though different headphones will naturally have different output at the same volume settings.) unbalanced signals will be internally converted to balanced. For IEM's (especially very sensitive pairs) that show up every little noise use the 3.5mm S-Balanced connection or add the 2.5mm iEMatch.
The xCan is Bluetooth 3.0 compatible and supports: As far as aptX HD, LDAC, LDHC are concerned, they cannot be supported on the current Bluetooth system used by iFi, therefore, it is not possible to perform a firmware upgrade but generally speaking, aptX offers very high quality as is, for CD standard and streaming audio. With AAC and aptX we cover the largest installed base of existing users. If you're still having issues, why not contact our support team? We are available Monday to Friday to give you a helping hand.
As the xCan connects via Bluetooth it needs a DAC, for this reason, an ES9023 Ess Sabre chip is used.
Under 1 Ohm for S-Balanced and under 2 Ohm for balanced.
All our headphone amplifiers are short circuit proof, meaning even a dead short will not damage them (but may trip protection circuitry). The following interactions of load and amplifier are absolutely not unique to iFi products, but apply universally to all and any amplifiers on the market, speaker amplifiers as much as headphone amplifiers. One consequence of using loads lower than the specified minimum is that maximum output power is reduced below specification, as the amplifier will limit the output current. Another is that for every halving of the load impedance distortion at low power levels is approximately tripled, with a given constant level. Modern amplifiers have a reasonably constant level of harmonic distortion nearly up to clipping, where the amplifier runs out of either current or voltage to drive the output. So lowering load impedance below the specified minimum increases distortion, possibly significantly, say we go from 16 Ohm to 4 Ohm distortion goes up approx. 9 times. How such increase in distortion is interpreted and heard subjectively is a different subject and if such increased distortion is audible if it remains substantially below the distortion of the transducer/headphone is yet another. A popular and fairly expensive IEM is officially listed by the manufacturer as 12.8Ohm @ 1kHz impedance and 112.8dB @ 1mW (normalised 132dB/1V). The real measured impedance is around 4Ohm below 300Hz with 7.5Ohm at 1kHz and 21-24Ohm from 7khz to around 10kHz. The real measured SPL at 1V is 141.5dB. Ignoring the sources of the discrepancies (which can be explained by different systems of rating rather than ill will and deception by the manufacturer as is commonly asserted in conspiracy theories), this IEM will be extremely challenging for any headphone amplifier and it will be hard to find a decent match. First, the very high sensitivity will emphasis hiss/noise. Typical high quality headphone amplifiers can be made to have around 3uV output noise, or -110dB/1V and > 3V output. With the 41.5dB/1V sensitivity such a noise levels means the noise itself is over 30dB in absolute level and clearly audible (and may even be intrusive) while the 3V output allow SPL of up to 150dB which is likely to damage ones hearing. The second factor is that the low impedance will increase the distortion of the driving amplifier in the bass nearly tenfold while in the mid-range it will be triple that specified for a 16 Ohm load. One option may be to manufacture a dedicated amplifier for such kind of IEM's, which should offer 0.3uV (-130dB/1V) output noise and a maximum of 0.3V output into 4 Ohm at low distortion and with low output impedance and has a maximum gain of approx. -10dB. As a 300 Ohm resistor produced more noise than -130dB/1V making such an amplifier would require very extreme design to achieve the required specifications and may still require cooling with liquid nitrogen to achieve the required specification. Instead at iFi we have specifically developed the iEMatch (integrated in some products and available standalone for others (and for those of our esteemed competitors) to address this problem and make such IEM's more compatible with more typical headphone amplifiers. Using iEMatch (which retails for less than 5% of the cost of the IEM in question) we achieve two things. The sensitivity of the IEM is reduced to 117.5dB/1V meaning noise levels of 3uV (-110dB/1V) now only produce an SPL of 7.5dB, at the very edge of audibility and maximum SPL is a still very unhealthy 127.5dB, which however is an order of magnitude less damaging to the human hearing than 151.5dB. Secondly, the minimum impedance seen by the amplifier is increased to 15.8Ohm, is 15.9Ohm at 1kHz and 16 Ohm at 7-10khz, meaning it will be easily driven within the specified performance by an amplifier rated for 16Ohm headphones, while the headphone itself sees a source impedance of less than 1 Ohm, comparable to a direct connection. So it is not required to use iEMatch to avoid damage to the xCan or to get sound when using ultra high sensitivity/ultra low impedance IEM's or headphones. However in the interest of hearing protection as well as pleasurable listening with music (lower noise, lower distortion) we strongly recommend that IEM's and headphones that combine very high sensitivity and very low impedances are used with an added iEMatch from iFi. This dramatically improves compatibility of these kind of IEM's with a wide range of amplifiers, not just iFi's, and usually costs a small fraction of the cost of the IEM or headphone in question.
Digital is a full signal at circa 2v whilst analogue via a smartphone is usually 0.75v. Hence the former much louder than the latter. Routed through the xCan or similar is the same for all.
The 'constant current drive grove' is a 21 Ohm resistor, as shown by JA measurements in Stereophile: 'Although Apogee describes the Groove's output impedance as variable and depending on the load, I measured a value of 21 ohms at all audio frequencies into loads ranging from 20 to 600 ohms (footnote 1).' Read more at??https://www.stereophile.com/content/apogee-electronics-groove-da-headphone-amplifier-measurements#TfvvQE60FmtqSm2H.99 And yes, a device with 21 Ohm constant output impedance can be connected to the xCan using 3.5mm connections without issues.
No, you can not use your Bluetooth headphones with the xDSD. This is because the xDSD does not have a Bluetooth transmitter, only a receiver.
The xDSD will draw the maximum safe current from any power supply unit (PSU), using the standard ID methods (Apple, Samsung, USB standard). Any standard-compliant PSU should signal it's current limit and the xDSD will then use this current limit. We do not recommend the use of non-standard compliant PSU's. The xDSD will charge at up to 1.1A and will, in addition, draw around 300mA to operate if the power supply is capable of providing this current level. If the PSU cannot be identified using standard methods, the xDSD will test the maximum current the supply can provide without dropping the output voltage below set thresholds, and then operate at this current limit. As an absolute maximum, around 1.4A may be expected if playing and charging a fully depleted battery.
The Bluetooth module consumes around 130mW continuous whilst the XMOS (USB) consumes around 350mW continuous at higher sample rates (less at 44.1k). The remainder of the xDSD (clock, reclocking, DAC, analogue stages and headphone amplifier) consumes around 740mW continuous, more with high power levels into low impedance headphones. The battery fitted is nominal 8360mW/H therefore: The playback times were rounded to the nearest hour after actual testing. All power consumption will fall within the 8360mW/H range and even a small amount of capacity degradation (10% in the first year) will not impact stated playback times.
Yes and no. For audio streams, Bluetooth V5.0 does not really offer material differences to 2.1 (neither do 3.0, 3.1, 4.2 etc). Almost all Bluetooth 5.0 improvements over 4.2 focus on low energy modes and nothing has been done for audio. Similar situations existed for earlier versions. To take advantage of the data-rate and range improvements, a new audio device profile (not A2DP) would be required.
The current generation xDSD does not include aptX HD, only 'general aptX'. This delivers CD-quality audio over Bluetooth and has comparably low latency, low enough to, for example, avoid lip-sync issues if watching TV. We also support aptX-Low Latency, customers tested this. If the source supports aptX-LL, and prefers it over standard aptX, it will be 32mS. https://en.wikipedia.org/wiki/AptX#aptX_Low_Latency
This is caused by a weak USB power source or low-quality cable. The xDSD will charge at 1.1A and requires extra power to operate. A 1.5A or higher current charger is recommended.
We have tested an xDSD on a Mac Mini OSX, the xDSD connects via AAC. There is no setup needed, the codec is selected automatically for the highest available sound quality. As far as we know on iPhone there is no special indication for AAC, on Mac it is visible in the audio device list.
There are 4 modes, these are: To change between Bluetooth and Wired Mode, when the unit is off you will need to hold the 'Rotary Knob' until the LED changes to either Green (Wired) or Blue (Bluetooth) and then release it. To engage 'Line out' mode, when the unit is powered off, hold (and remain holding) the 'Bluetooth/Gear' icon then select either 'Wireless' or 'Wired' by press holding the 'rotary knob' UNTIL the LED changes to either BLUE (Wireless) or GREEN 'Wired' then let go. Then let go of the 'Bluetooth/Gear' icon. This will active Line out mode. To remove Line out mode, just repeat the same process as if you would be activating it.
Plug into external power and then hold down the power button for over 10 seconds. If for some reason the unit is stuck in an unintended state, that should reset it, kind of like the reset button combo on phones.
This LED indicates audio format (PCM/DSD/MQA) and its sampling frequency.
Potential solution: If the rotary knob has any colour, please rotate it completely counterclockwise until the colour is no longer present. After doing this you should see the input LED blinking blue and red, which is now ready to reconnect. Another solution: If after following this first solution you are still unable to connect your device to your xDSD then another solution would be to drain the battery of your xDSD. By doing this it performs a 'soft reset' which allows for new connections to be made.
Yes, you can charge and play at the same time. If you actually have the xDSD permanently attached to USB power it will enter 'desktop' mode after the first time the battery is charged fully and the unit is turned off. In desktop mode, the battery is charged to ~ 70% and kept in maintenance mode. This state is maintained until the xDSD enters battery power mode (unplugged or turned on battery).
Yes, we do support quick-charge if suitable chargers are attached. A full charge from flat is approx. 3 hours, providing it has a minimum of 5V/1.5A or higher rated charger.
The noise is for the first few seconds when connecting the charger always happens, it will be silent afterwards. If a weak USB power source or low-quality cable is added then noise may be present. The xDSD will charge at 1.1A and requires extra power to operate. A 1.5A or higher current charger is recommended.
Yes and no. For audio streams BT 5.0 does not really offer material differences to 2.1 (neither do 3.0, 3.1, 4.2 etc). Almost all BT 5.0 improvements over 4.2 focus on low energy modes and IoT (Internet of Things / home automation etc) and nothing has been done for audio. Similar situations existed for earlier versions. To take advantage of the data-rate and range improvements a new audio device profile (not A2DP) would be required.
The xDSD runs 'AB' in fact, almost all modern audio electronics run in Class AB, even most so-called 'Class A' becomes 'AB' for some combinations of load and signal. Generally, we call 'A' when normal listening levels with all kinds of loads happen in Class A (e.g. iCan Pro). This 'Class A' means a lot of power is wasted, Class A means the maximum power we want in Class A is permanently wasted as heat but even the xDSD operated in Class A for small signals. The specific IC we use as headphone buffer uses internal circuitry that reduces the distortion caused by the Class A to Class AB transition to very low levels, so kind of Class A sound with Class AB power consumption, ideal for battery-powered gear.
Yes, you can!
Yes you can. Plug the xDSD gryphon into a PC via USB and use the Dual Port setting. This will charge the battery to 3.88v and it will stabilize once charged, so the device will stay powered on without using the battery
Yes, when not in Line-In mode, 4.4mm Pentaconn and 3.5mm sockets on both sides output audio simutaneously. However, iEMatch is only applies to the sockets on the side with the Power/Multi-function knob.
Yes. The Gryphon supports coax via 3.5mm adapter, and Toslink optical via mini-Toslink adapter on the same physical socket.
xDSD Gryphon supports advanced features including: Ultra-Res (PCM 32/768, 24?/192, DSD512, MQA decoder 384) Advanced 96kHz Bluetooth 5.1 QCC5100 chipset, HD Bluetooth formats Purewave balanced design (from Neo and Diablo), lower noise/THD? Filters ' GTO, STD, BP iEMatch Balanced 4.4mm outputs Dual/multiple headphones simultaneously Multiple input selections: Bluetooth, USB, S/PDIF (coax & optical), LineMultiple output selections: 4.4mm BAL, and 3.5mm S-BAL
With USB power unplugged, hold down the XSpace/XBass button while turning on the unit will factory reset the device.
For firmware v1.07 1. Hold power down in an attempt to turn it on (you would have done that by now). 2. Unit may feel slightly warmer than other units. 3. Let it sit for a day or two. It's actually discharging. 4. After it's completely discharged, unit feels cold again, it can be turned on and charged again normally. 5. Please update to latest firmware.
iEMatch is our proprietary tech to reduce hiss from IEMs due to their having too much gain. The attenuation is up to -12dB.
YES!
Analogue so you can enjoy the best sound quality because digital volume controls cut the resolution by up to 1.5 Bits. The neat trick in the Gryphon is that it operates in the analogue but the controls are in the digital domain so you can have the best of both worlds!
XBass II denotes you have the option to perfectly fine-tune the bass frequency response. Bass ' is low bass and Presence is mid-bass. Please adjust as you like to get the ideal bass response for you.
A: On the 3.5mm HP output, use at a setting between 0dB to -3dB for lowest distortion. Lowest distortion is at 0dB to -3dB, depending on how high the digital modulation is. At less digital modulation (typical music etc.) 0dB setting has lower THD. At 0.5 FFS it's 0.00143% on 0dB setting. At 1.0 FFS it's 0.0036% on -3dB setting.
There's not a need to do so because the'SilentLine' tech means the OLED and its power supply are de-coupled from the main PCB. Therefore there is next to nil noise contamination.
USB-C with 5v. High powered chargers are supported.
DXD is Digital eXtreme Definition and was created for editing DSD music. Since the 1bit operation of DSD is not suitable for music editing, most of them have to use PCM (Pulse-code modulation), so DXD is a PCM format for music recording and editing DSD. The specification is 24bit/352.8kHz PCM signal, in which 24bit has 8bit more data than the 16bit sampled by CD, and the sampling rate of 352.8kHz is 8 times that of CD 44.1kHz, and the data transfer rate is 8.4672 Mb/s, which is 3 times that of DSD64 commonly used in SACD, and can provide better analysis than CD and SACD. This provides better analysis and dynamics than CD and SACD, representing one of the highest standards of sound quality for studio production. DXD was originally designed for Merging Pyramix's music production workstation and was introduced in 2004 along with the Sphynx 2 AD/DA converter used in the workstation. By using DXD to edit music, only one final conversion to DSD is required before producing SACD, which greatly reduces the high frequency noise above 20kHz generated during DSD conversion and preserves better sound quality. The DSD format itself is prone to noise in the high frequencies, and each DSD conversion introduces more noise. DXD can limit the conversion to one edit, which helps reduce the sound quality loss of DSD/SACD.
xDSD Gryphon is our newest portable DAC that supports multiple inputs and outputs. For additional details, please check out Gryphon webpage on our website.
8 hours typically
The device should work perfectly fine out of the box, but if you want the newest features or ever start to experience issues, we always suggest updating to see if that fixes the problem. How to update on PC:??https://www.youtube.com/watch?v=xfv32H0QoIM How to update on Mac:??https://www.youtube.com/watch?v=F-JVJhCMq8U Bluetooth update for Zen Air Blue on Android:??https://www.youtube.com/watch?v=ZFB91czZhCs If you ever need help with an issue or an update, please feel free to submit a support ticket with us.
We wanted to create a high quality DAC for customers on a budget, so a power supply is not included, but you can power the device via the USB port. To fully drive the device, you can use any 5v power supply.
The DAC powers on upon either connecting it via USB to a PC or laptop, or by providing 5V from an external power supply within the required specs.
Zen Air DAC quick set-up guide:??http://www.youtube.com/watch?v=3tYhTrJ77Dg Zen Air Blue quick set-up guide:??https://www.youtube.com/watch?v=50B1RvShUF0
To turn off the LED display, short press the pairing button. Pressing it again will turn the display back on.
Yes, it's perfectly fine to have the DAC constantly powered on. It draws little power and it will not damage the DAC.
Yes! You can verify your student status here:??https://downloads.ifi-audio.com/about-ifi/student-discount/
This LED indicates audio format (PCM/DSD/MQA) and its sampling frequency. For more information on what each color means, please use our LED Calculator tool:??https://downloads.ifi-audio.com/home/led-calculator/
DAC stands for Digital to Analog Converter, and it does exactly that. It converts a digital signal to an analog signal. Any device that outputs a digital sound, be it a CD or Blu-ray player, digital TV box, games console or portable music player, will need a DAC to convert its audio to an analogue signal. Your TV, phone, and laptop already have a built in DAC that converts digital sound to an analog sound we can hear. Using a high-end DAC will result in the highest quality conversion between digital to analog, which means you're getting the audio the way it was meant to be heard.
MQA''?Master Quality Authenticated??is an award-winning audio technology that enables music fans to stream the original master recording into their home, car or their mobile. With the ZEN Air series of DACs, you can listen to MQA straight out of the box. Simply?? connect to Tidal Masters and check the option to stream MQA. You can find more information about MQA here: https://www.mqa.co.uk/newsroom/faqs/what-is-mqa
Yes, the Zen DAC will work if powered separately by a 5v power supply. Connect the DAC to the USB port on the front of the PS5. Depending on which Zen DAC you have, the USB ports on the back of the PS5 may also be used. The Zen DAC v1 will work with XMOS firmware. The Zen DAC v2 and Zen Air DAC will work with firmware 7.4.
The Zen Dac will need to be connected via USB to the PC first to establish a connection, then a 5v power supply can be added.
The ZEN Air DAC and ZEN DAC V1/V2 will receive power from a 5V PSU, however to activate these DACs, you must make a successful USB connection. You can identify that the DAC is being powered by the PSU because the 'external power' LED next to the power input will activate.
With bluetooth firmware version 2.30, we have added the feature to reset bluetooth pairings. Hold down the pairing button on the front of the device for 15 seconds, if done correctly you should see the LEDs change color.
To turn off the voice announcements, short press the pairing button two times. To turn it back on, double press the button again.
Yes, there is a sleep function, if it stays in pairing mode without a connection for 5 minutes it will go into to sleep mode. This also stops the LED of the pairing mode being too bright, from blinking red/blue, to calm pulsing blue. You will need to press the button to wake it up however.
First, we have a new update for the Zen Blue, please find instructions here. Second, ??LDAC and the QCC chips are constantly under investigation by Sony to strive for future improvements,
There is new pre-loaded firmware for the ZEN Blue V2. This improves the signal for best LDAC operation. The original ZEN Blue can be updated from our website. Please go to our Download Hub to find out more.
Yes, with the rear switch in analogue position both analogue outputs are active.
Yes, it allows the flexibility to connect more than one DAC or system so that you don't need to unplug a cable while listening to the other one.
The main difference is that the internal clock has been upgraded with low phase noise for even lower jitter. This results in even higher resolution sound quality. There is also an additional larger antenna for extended reception.
Game mode caters directly to the needs of those constantly seeking any advantage they can get in competitive play. Gamers understand the importance of hearing even the faintest sounds that can give away an opponent's position or signal an approaching threat. Game mode's EQ enhances this aspect of gameplay by sharpening subtle, low-level sound effects, ensuring that every sound becomes more distinct and easier to pinpoint. Whether navigating through intense firefights, or sneaking up on enemies, this heightened audio sensitivity allows you to stay one step ahead of opponents. Movie Mode is an EQ specifically designed to address the clarity of dialogue in streamed content, enabling you to fully immerse in the narrative without straining to hear or decipher what is being said.
Yes, the ZEN CAN Signature 6XX can work as a fully balanced pre-amp for such products.
One press of the distinctively labeled button on the ZEN CAN Signature 6XX's front panel is all it takes. An active white LED nearby indicates that ActivEQ is engaged.
Various LEDs on the ZEN CAN Signature 6XX's front panel will be engaged.
Internal components make all the difference. The ZEN CAN Signature 6XX incorporates many boutique parts taken directly from our flagship product, a ??3,000 Pro iCAN.
Internal components make all the difference. The ZEN CAN Signature HFM incorporates many boutique parts taken directly from our flagship product, a ??3,000 Pro iCAN.
Although the ZEN CAN is a great bang for your buck, the ZEN CAN Signature 6XX is audibly better and has ActiveEQ which is not present in its basic version.
The ZEN CAN Signature 6XX features a switchable ActivEQ equalizer specifically designed to get the most out of the HD 6XX. One button press is all it takes to calibrate the amp's signal to complement the HD 6XX's frequency response characteristics, which results in its performance tailored to precisely fit these headphones like a made-to-measure suit. What's more, ActivEQ does this without adding any of the distortion commonly associated with passive equalization circuits.
Although the ZEN CAN Signature 6XX is visually similar to our ZEN CAN and both these products share the same core platform, the former has it pushed significantly further to achieve better sound performance.
XSpace is our own in-house developed fully analogue headphone spatialiser, meant to widen audible imaging of headphones connected to the ZEN CAN Signature 6XX.
This button cycles through an amplifier's gain and the higher its setting in dB is, the louder sound level will be the result.
ActivEQ, combines active and passive components to create a specific EQ curve to suit a particular pair of headphones, performed in the analogue domain without a hint of additional noise or distortion.
If possible, please use the ZEN CAN Signature 6XX's balanced 4.4mm inputs/outputs as only those engage its fully balanced circuitry.
Although any charger capable of providing 5V will do, we highly recommend our own ultra-quiet iPower X power supply.
Please connect your headphones of choice and set the PowerMatch gain to have the ZEN CAN Signature 6XX's volume knob around 12 o'clock at normal listening levels.
This knob controls volume level of connected headphones, external power amplifiers or speaker sets.
Due to the ZEN CAN Signature 6XX's fully balanced topology we strongly advise to use its balanced 4.4mm Pentaconn output, but its 6.3mm single-ended socket is also perfectly fine.
It will, as long as frequency response of such headphones is at least partially similar to the HD 6XX. That's the reason why Sennheiser's models HD 600/650 work really well with ActivEQ!
Yes, the ZEN CAN Signature HFM can work as a fully balanced pre-amp for such products.
One press of the distinctively labeled button on the ZEN CAN Signature HFM's front panel is all it takes. An active white LED nearby indicates that ActivEQ is engaged.
Various LEDs on the ZEN CAN Signature HFM's front panel will be engaged.
Although the ZEN CAN is a great bang for your buck, the ZEN CAN Signature HFM is audibly better and has ActiveEQ which is not present in its basic version.
The ZEN CAN Signature HFM features a switchable ActivEQ equalizer specifically designed to get the most out of HIFIMAN headphones. One button press is all it takes to calibrate the amp's signal to complement HIFIMAN's frequency response characteristics, which results in its performance tailored to precisely fit these headphones like a made-to-measure suit. What's more, ActivEQ does this without adding any of the distortion commonly associated with passive equalization circuits.
Although the ZEN CAN Signature HFM is visually similar to our ZEN CAN and both these products share the same core platform, the former has it pushed significantly further to achieve better sound performance and features distinctive sonic tailoring, the ActivEQ.
XSpace is our own in-house developed fully analogue headphone spatialiser, meant to widen audible imaging of headphones connected to the ZEN CAN Signature HFM.
The ZEN CAN Signature HFM is a fully balanced desktop headphone amp generously loaded with boutique parts and a very special analogue filter 'ActivEQ' tailored specifically for HIFIMAN headphones.
If possible, please use the ZEN CAN Signature HFM's balanced 4.4mm Pentaconn inputs/outputs as only those engage its fully balanced circuitry.
Please connect your headphones of choice and set the PowerMatch gain to have the ZEN CAN Signature HFM's volume knob around 12 o'clock at normal listening levels.
Due to the ZEN CAN Signature HFM's fully balanced topology we strongly advise to use its balanced 4.4mm Pentaconn output, but its 6.3mm single-ended socket is also perfectly fine.
It will, as long as frequency response of such headphones is at least partially similar to HIFIMAN headphones.?? Please see this compatibility chart for details of how well the ZEN CAN Signature HFM works with the range of HIFIMAN headphones.
The ZEN DAC sports 4.4mm balanced outputs at both the front (for headphones) and back (for output into another amplifier via 4.4mm to dual XLR). We've used a balanced design for the analogue stage' rare and not found at this price point. This makes it pretty special for the best sound quality. A balanced cable has the same left + and right +, but it has a separate left 'and right -, with totally separate wiring.'Crosstalk' is negligible. Therefore, a balanced connection is the superior headphone wiring configuration.
Yes, but you'll have different power levels/impedances etc.
This was a mistake on our end, we have amended the online manual and also physical manual. The Zen DAC doesn't come with a power supply, only a USB cable that takes power from a USB source like a computer. Please see our new user manual.
Yes, they are the only sockets required to send audio signal in its analogue form from the ZEN DAC Signature to a different product.
Yes, you can connect your balanced stereo power amp to a 4.4mm out, and a subwoofer to RCAs.
Although both options are perfectly fine, in order to maximize sound performance, we advise to power the ZEN DAC Signature via our iPower X power supply.
The ZEN DAC Signature powers on upon either connecting it via USB to a PAC/laptop, or providing 5V from an external power supply within required specs.
Internal components make all the difference. The ZEN DAC Signature incorporates many boutique parts taken directly from our flagship product, the ??3,000 Pro iDSD.
Although the ZEN DAC is gives you great bang for your buck, the ZEN DAC Signature is audibly better. At least that's what our customers say on the matter.
Sound quality difference isn't staggering, but audible enough to notice and appreciate it.
Yes, it's perfectly fine to have the ZEN DAC Signature constantly powered.
The way that the ZEN DAC Signature processes digital data is quite unusual and highly sophisticated. It eliminates jitter from the source via the separate Global Master Timing (GMT) circuit inherited from our high-end audio parent company AMR, and then the Burr-Brown's D/A chip outputs the converted analogue signal to the end-to-end balanced analogue circuit with custom-designed premium OV2637A op-amp.
It's entirely up to you, however USB3.0 cables provide better mechanical connection and that's what we suggest to use.
TI low-noise power supply ICs, well-designed ground planes for the PCB, ELNA Silmic II, Panasonic OS-CON and ECPU, Taiyo Yuden and muRata AND TDK C0G capacitors.
The ZEN DAC Signature is a fully balanced desktop DAC/pre-amp product generously loaded with boutique parts and also a MQA renderer that goes up to 32bit/384kHz PCM and DSD256.
RCA/4.4mm Pentaconn line level outputs on the ZEN DAC Signature's rear panel can be set either as fixed or variable via a switch located there. Upon selecting the former option the product works only as a DAC and has its front knob disengaged, whereas the variable mode turns it into a DAC/pre-amplifier device, which can also regulate volume of connected power amps.
The ZEN DAC Signature is a fully balanced product, so its RCAs are single-ended, whereas the 4.4mm Pentaconn socket is its balanced output.
Yes, they are the only sockets required to send audio signal in its analogue form from the ZEN DAC Signature V2 to a different product.
Although both options are perfectly fine, in order to maximize sound performance, we advise to power the ZEN DAC Signature V2 via our iPower X power supply.
The ZEN DAC Signature V2 powers on upon either connecting it via USB to a PAC/laptop, or providing 5V from an external power supply within required specs.
Internal components make all the difference. The ZEN DAC Signature V2 incorporates many boutique parts taken directly from our flagship product, the ??3,000 Pro iDSD.
It's entirely up to you, however, USB3.0 cables provide better mechanical connection and that's what we suggest to use.
The ZEN DAC Signature V2 is a upgraded version of the original. It now has a 16 core XMOS chip, rather than 8 core. This means faster processing and the ability to be a MQA decoder rather than just a renderer.?? It also has an upgraded crystal clock with 20dB better performance. Both products are similar to our ZEN DAC and both these products share the same core platform, but the Signature range has it pushed significantly further to achieve better sound performance.
Although the ZEN DAC is gives you great bang for your buck, the ZEN DAC Signature V2 is audibly better. At least that's what our customers say on the matter.
Yes, it's perfectly fine to have the ZEN DAC Signature V2 constantly powered.
The way that the ZEN DAC Signature V2 processes digital data is quite unusual and highly sophisticated. It eliminates jitter from the source via the separate Global Master Timing (GMT) circuit inherited from our high-end audio parent company AMR, and then the Burr-Brown's D/A chip outputs the converted analogue signal to the end-to-end balanced analogue circuit with custom-designed premium OV2637A op-amp.
The ZEN DAC Signature V2 is a fully balanced desktop DAC/pre-amp product generously loaded with boutique parts and also a MQA decoder that goes up to 32bit/384kHz PCM and DSD256.
This LED indicates audio format (PCM/DSD/MQA) and its sampling frequency.
RCA/4.4mm line level outputs on the ZEN DAC Signature V2's rear panel can be set either as fixed or variable via a switch located there. Upon selecting the former option the product works only as a DAC and has its front knob disengaged, whereas the variable mode turns it into a DAC/pre-amplifier device, which can also regulate volume of connected power amps.
The ZEN DAC Signature V2 is a fully balanced product, so its RCAs are single-ended, whereas the 4.4mm socket is its balanced output.
Yes. They are the only sockets required to send an audio signal in its analogue form from the ZEN DAC V2 to a different product such as the ZEN CAN.
Yes, you can connect your balanced stereo power amp to a 4.4mm Pentaconn out, and a subwoofer to the RCAs.
Although both options are perfectly fine, in order to maximize sound performance, we advise powering the ZEN DAC V2 with our iPowerX power supply.
The ZEN DAC V2 powers on upon either connecting it via USB to a PC or laptop, or by providing 5V from an external power supply within the required specs.
Yes, it's perfectly fine to have the ZEN DAC V2 constantly powered on.
The way that the ZEN DAC V2 processes digital data is quite unusual and highly sophisticated. It eliminates jitter from the source via the separate Global Master Timing (GMT) circuit inherited from our high-end audio parent company AMR, and then the Burr-Brown's D/A chip outputs the converted analogue signal to the end-to-end balanced analogue circuit with custom-designed premium OV2637 op-amp.
The rear panel outputs can be set either as fixed or variable via a switch located there. Upon selecting the fixed output the product works only as a DAC. Selecting variable mode turns it into a DAC/pre-amplifier device, which can also regulate volume of connected power amps. The front panel outputs are controlled by the volume control.
The ZEN DAC V2 is a fully balanced product, so its RCAs are single-ended, whereas the 4.4mm socket is its balanced output.
The Zen Phono MCU is designed to switch out of the signal path after a few seconds to improve the sound quality. This means when turning the Zen Phono off we need to press the on/off button twice . First time to wake the MCU, then wait a few seconds and press a second time to turn the unit off.
Yes. When you plug a USB source into the rear USB socket, the ZEN Stream will automatically sync its contents, and you can view them on the iFi portal http://ifi.local under Sources.
Yes. Connecting your phone or portable device with the ZEN Stream's hotspot can be useful for yachts, motorhomes and other portable setups. This allows you to control a connected USB drive to playback its contents. Your portable device can also playback stored music files this way. If using a streaming service such as Tidal, Qobuz, Spotify etc. and sending it to the DAC via the Stream's hotspot connection, the portable device is forced to use mobile data and data charges may apply ' check with your mobile service provider.
If your DAC can be run from USB power, yes. The ZEN Stream incorporates Active Noise Cancellation (ANC) on the 5v USB power output to provide the DAC, and/or USB music source, with noise-free power. The ZEN Dac, ZEN Dac V2, and the iFi Neo iDSD can both be run from this high-quality power source if appropriate.
No, you can use your phone network to connect to the ZEN Stream using its built-in Wi-Fi hotspot facility, eliminating the need for a computer or Wi-Fi router.
No. If the USB output is used the coaxial output switches off. If no USB DAC is plugged in, the coaxial output switches on. The outputs can also be managed on the http://ifi.local portal under Settings 'Playback Options' Audio Output.
Yes. Alternatively, use the Wi-Fi hotspot. Please see the FAQ 'Do I need a Wi-Fi network router'?
View the iFi portal http://ifi.local under Settings 'System' System Updates. Here you can check if there is updated firmware available. The current version on your ZEN Stream is shown under System Version on the same page.
There are two different volume options available for the Zen Stream: 1. When the 'Volume Option' is set to 'Software', the digital volume can be adjusted. The UI volume (digital volume) and the external USB DAC volume (analog volume) can be adjusted separately. 2. When the 'Volume Option' is set to 'None', the digital volume is fixed at the maximum level, and only the external USB DAC volume can be adjusted.
If your ZEN Stream is near a router/modem, you can use an Ethernet connection. This may give you a more secure connection to your internet service.
The ZEN Stream has two outputs, USB and S/PDIF coaxial. The coaxial output complies with S/PDIF standards with a maximum frequency of 192kHz for existing and older DACs. You can also play DSD files up to DSD128 through the coaxial output ' it converts them to PCM 192kHz.
This switch allows you to turn off facilities not in use for lower internal noise and system efficiency. All In One (AIO) lets you use all the capabilities of the ZEN Stream if required including initial setup. For example, using the Tidal position (switch position 3) allows only streaming from Tidal, and all other facilities are switched off. More specific preferences can be set on the http://ifi.local portal under Settings ' Sources. In order to view this page, the switch needs to be on AIO.
The ZEN Stream acts as a wireless or wired connection between an internet-enabled or Wi-Fi network-connected device and a DAC. It's a dedicated wireless transport device that allows the use of music streaming services such as Tidal, Qobuz, Spotify etc. and DLNA clients such as Audirvana, Roon, JRiver etc. Please see the Lowdown for more examples.
The ZEN Stream will also connect to and operate NAS drives on the same wi-fi network, and USB storage devices plugged into the rear USB 3.0 socket. These are controlled by the dedicated iFi portal http://ifi.local accessible on your device's browser.
Please make sure the System Mode Selector is set to 1, All In One mode to access ifi.local.
The most common use is to employ a sonically high-quality wireless connection from a portable Wi-Fi and/or Internet-enabled device to a DAC. Eg. iPhone/iPad/Android device 'wi-fi network router' ZEN Stream ' USB DAC such as the ZEN DAC V2.
Although labeled differently, both USB ports are connected to the same hub and will provide 5v to a connected DAC.
You will need to use your computer with the ZEN Stream for playback while using these DLNA media players.
In some situations, electrical isolation between a computer and a USB DAC may be beneficial. This can eliminate ground loops and avoid power supply noise originating in the computer power supply from entering sensitive DAC circuitry. As the ZEN Stream is a wi-fi device it will achieve this. The iFi iGalvanic3.0 also does this.
The ZEN Stream box contains: 1 x ZEN Stream 1 x Ethernet cable 1 x Plastic screwdriver 1 x Power adapter 1 x Antenna 1 x Instruction card 1 x User manual
150 ohm.?
The iCan Pro output impedance varies very little with different modes.
Question: Is the output OK to use with other devices? Answer: It is for the iESL Pro only, it is not digital, it is matched to iESL Pro.
The micro iUSB 3.0 reclocks the USB data stream. It is not straightforward to measure jitter at 480mHz (which has a cycle time of around 2nS). However, any USB cable attached to the miro iUSB 3.0 output will likely add higher levels of jitter than is contained in the micro iUSB 3.0 output, which is locked to a local crystal clock which should have, however, the jitter from the micro iUSB 3.0 does not translate directly into jitter on the DAC side, this is down to the USB system in the DAC. Again, it is safe to say that the micro iUSB 3.0 will have a much lower contribution to jitter from the USB system than how the USB system in the DAC (or USB to SPDIF converter) is designed.
Chromebook and similar netbooks/tablets such as Microsoft Surface are somewhat vague about whether or not they are USB Audio Class 2.0 compliant. We have not tested the Chromebook 'it is best you confirm with Google regarding'USB Audio Class 2.0.'But according to: http://gigaom.com/2012/08/21/update-makes-chromebo' 'Support for standard USB audio devices and wireless gamepads' suggests that USB Audio Class 2.0 in built-in.
ASIO native and DoP are two different ways to send DSD data through the USB audio subsystem. Both require hardware support (in the DAC) and software support (in the playback software). The key difference is that ASIO has it's own distinct protocol that packs 32 bit worth of DSD data into a PCM sample but support even on Windows is very limited and does mostly not exist on Apple& Linux. Meanwhile, DoP is totally platform/device/driver agnostic but can only transmit 16 Bit worth of DSD data per 32 Bit PCM sample (the rest is used up by the protocol marker and other overheads). So ASIO 88.2KHz native can transmit DSD64 while on DoP it needs 176.4KHz to transmit DSD data. Other than that fundamentally the two systems are very similar. Neither transcodes DSD into anything else, both use PCM Packets as'transport' and both re-assemble the original DSD datastream completely transparent and bit-perfect before it is sent to the DAC. Specifically using DoP, DSD is sent as a 16-bit chunk per PCM sample. So DSD(64) requires 4 PCM samples to hold 64 DSD bits and thus it needs to be sent as 176.4kHz PCM. And thus DSD128 via DoP requires 352.8kHz PCM and DSD256 via DoP requires 705.6kHz PCM. Only via ASIO (native) is DSD sent at the DSD sample rate, though under the bonnet it is sent in effect as 32 Bit per PCM sample, so 705.6Khz allows 22.57M DSD sample rate (DSD512). The key difference between ASIO native and DoP is the way DSD signals are identified and how many bits are available per sample. ASIO has a mechanism separate from the data to signal DSD and uses 32 bit per PCM sample. For DoP the DSD marker is embedded in the PCM data stream and only 24 Bits are used of which 8 Bits are used as the DSD marker.
We generally do not provide detailed information on the design specifics of our products. Which particular chip is used is only a small percentage of the final sound quality. We are only able to disclose which chip is used when its officially stated in forums / technical papers or via another means. If you require the information for servicing, please instead work with our Technical Support Department to provide you with replacement PCB and/or replacements parts directly from iFi.
USB audio itself supports up to 32 bit integer or floating-point, however, floating-point is generally not supported. 64 bit floating point audio is not supported by USB audio at all. Conversion happens inside the computer. However, actual music recordings are currently only available in up to 24 bit fixed-width (integer) audio and no electronics or microphones exit that comes close to 24 bit equivalent signal/noise ratio or dynamic range, in fact, we know of no microphone/pre-amplifier system that even approaches 20-Bit performance at normal sound pressure level's from a jazz band or orchestra. Also, these 24-bit music files should be left as this when playing back on computers, so any debate about 32 bit, 32-bit floating point or 64 bit anything and what happens to them is completely academic.
There is no simple 'one size fits all' number that makes sense. For example, 44.1kHz Bitperfect will have -3dB at 18kHz, 96kHz standard filter will have at 48kHz, 384kHz will have a -3dB of around 80kHz due to the analogue filter, rather than the 192kHz -3dB expected from the Bitperfect filter. So the answer to the question 'What is the frequency response'? is, it depends upon sample rate and filter chosen and can range from 18kHz (44.1kHz bit-perfect) to 80kHz (-3dB sample rate above 176.4kHz).
The exact timing of the refund arriving in your account depends on the payment method you used. This usually takes 3-5 business days from the date the order is cancelled. Please note that the refund is not always shown in the statement.
The micro iDSD Signature is a micro iDSD BL, but internally streamlined and optimized, and externally equipped with several QOL (quality of life) improvements to be more user-friendly.
Simple, when iFi supplied EL84X are fitted the jumper must be closed. If using most current production EL84 as replacement the jumper is inconsequential. Certain specific NOS EL84's (some batches of Mullard are known) have an undocumented connection to a pin marked'NC'(no connection) and thus this pin must be disconnected for these specific tubes. Many EL84 amplifiers connect this pin by default (which can lead to damaged tubes/amplifiers if the a NOS EL84 is fitted that has this internal connection. Some modern EL84 variants do not perform correctly if the jumper is open. Bottom line, EL84 with suffix (M, X etc.) need the jumper closed' straight modern EL84 do not care 'new old stock EL84 must have the jumper open. We do however not recommend modern manufacture EL84 (including those re-issued under old brand names) as a replacement for the EL84X, please get original replacement sets from iFi if needed. Most NOS EL84 from reputable manufacturers will work in the Stereo 50 however the Stereo 50 runs tubes quite hard, life expectancy is estimated 5,000 hours or less than 2 years with 8 hours playtime per day and the ECF82 has a greater impact on sound quality, so using up rare and expensive NOS EL84 in the Stereo 50 may not be the best choice. For people who would like to'tube roll' we recommend trying the ECF82 first and leaving the EL84X alone. Please note, tube rolling does void the warranty.
Typically, those albums are encoded as a 48kHz MQA file. Meaning TIDAL will perform the decoding to a 96kHz audio stream with the MQA renderer handling the further decoding, which has nothing to do with our products. This is how MQA & Tidal works. To test that the actual audio stream in TIDAL is 48kHz, please select 'MQA pass-through'. It is 96kHz if TIDAL decodes MQA, it is 48kHz on MQA pass-through. Question:??But I thought the file was 192kHz remastered? Answer: Yes, meaning it was MQA encoded from a 192kHz remaster. Our MQA implementation handles up to 384kHz MQA sources correctly according to MQA specifications.
Please use a genuine Apple camera connection kit (see other FAQ) then switch on the battery first (it will activate and avoid pulling power from the Apple device) then connect it to the Apple device. Otherwise the iFi device is trying to draw power from the source (thinking its a computer) which in this case, the Apple device will then say no (ie: this accessory is not supported).
In a corporate environment, you can use Windows 10 latest revision, as this has a built-in USB Audio Class 2 driver. However, yes, there are some limits. Basically, the limit is DoP for DSD, so for our current firmware, 5.2 > DSD256 limit, 5.3 > DSD128 limit.
We do not support virtual machines. If the driver installs correctly on the machine running Windows it means the hardware and software works. Any support regarding the use of virtual machines on Windows needs to come from the software vendor (Microsoft). This is because virtual machines can affect video and audio hence we do not support them.
A customer reported using this cable, and after testing we can confirm that it works: https://www.apple.com/shop/product/HKKP2ZM/A/belkin-lightning-audio-charge-rockstar
'Code 10' in this case means there is no driver (the device cannot start without a driver, naturally). In cases like this: 1) Make sure to run the set-up file with administrator rights (right-click 'select run as administrator). 2) Make sure to temporarily turn off any anti-virus software for the installation. 3) Server versions of Windows often have elevated levels of security preventing the installation of drivers etc., turn these to normal during installation. 4) If none of these work, verify the driver install and device work on a'normal' PC, and then seek support from Microsoft.
Androids require USB 2.0 OTG (on the go) because phones need to be a 'device' if attached to a host, but they must be 'host' if attached to a device. Basically, phones play for both teams. There are a lot of Android machines from different manufacturers, so it really does depend on the model, manufacturer, and even OS version installed. Here are the basic requirements: The links below are great resources which are very useful for Android users, and are being constantly updated. Please check them out: http://www.extreamsd.com/USBAudioRecorderPRO/ http://www.head-fi.org/t/595071/android-phones-and-usb-dacs/1140 In short, if it conforms to the above specifications, then it will work with the iFi device or any other USB Audio Class 2.0 compatible DAC for that matter. Please see here for more Android reading: https://downloads.ifi-audio.com/home/computer-audio-tutorial/android/
What we have done is disable the USB device volume control (which uses DSP in the XMOS system to control volume). OSX 'obeys' this while other operating systems have additional DSP (usually optimised for low CPU load over sound quality) that adds extra volume controls that are preferably left alone. So we have not 'locked away' anything from Mac users, instead, OSX avoids an extra layer of DSP (as does Windows in exclusive mode WASAPI). If volume/balance is adjusted, it invariably causes loss of quality and it is not recommended to use these functions. Most speakers and headphones have sufficient channel matching that balance adjustment is not required.
(From MQA) ' Whether the decoding and rendering takes place in the same device, or the process is split between a software application and a rendering device, the result will yield the full, original studio sound. For more information, please contact MQA directly at:??[email??protected]
No virus is contained within our drivers. This is a false positive. Please disable your anti-virus for 5 minutes whilst you install the driver. In fact here is a customer contacting Windows to verify the drivers security: https://www.head-fi.org/threads/ifi-driver-software-benevolent-or-malish.862118/page-2 Quote:??'My submission seems to have had the required effect. The Windows Defender Team have confirmed that the file is not malware AND Microsoft Security Essentials is no longer complaining about the file. Hurrah!'
I noticed that I'm not able to use any of these 48000hz DSD sampling rates: But the corresponding 44100Hz rates are supported. I'm using the 2.26 driver on Windows 7 Ultimate. These sampling rates (except DSD512, which doesn't work under DoP) work using DoP on Mac, and natively on Linux. Is there a reason the Windows driver has been crippled to not provide the rates detailed above? Answer: Officially DSD uses 44.1kHz based sample rates only. The ASIO driver supplied by XMOS and used by iFi follows this convention and only supports 44.1kHz based DSD. We have raised 48kHz support as a feature request, however, so far this has not been implemented. If 48/96/192/384kHz PCM audio is to be used with conversion to DSD, please use either DoP (limited to 24.576MHz DSD / DSD256) or use software that converts to standard DSD frequencies if using ASIO.
Please untick the 'Integer Mode'. Normally this solves the issue as OSX supports of 'Integer Mode' between versions is on and off. Also the 'Direct Mode' superseded the 'Integer Mode' anyway. But on some OSX machines, one may need to untick the 'Direct Mode' too.
The following is from Audirvana's designer in relation to DSD/DoP:'To listen to DSD files from a DAC, Audirvana Plus transmits the DSD files to the DAC using the 'DSD over PCM' protocol that requires markers, and using this protocol requires twice the bandwidth over the native DSD signal only. In order to get the best possible result, you need to use Audirvana Plus with a Native DSD DAC. However, if your DAC is not DSD compatible, Audirvana Plus will perform downsampling to PCM using a very high quality 64 bit algorithm. Please note : The 'DSD over PCM' protocol requires markers to identify the signal as DSD, so using this protocol requires double the bandwidth compared to the native DSD signal alone. This can reduce the maximum DSD frequency that can be used with DAC by half compared to the frequency that can be reached under Windows. This may explain why a DSD 512 DAC can stop at DSD 256, and DSD256 DAC can stop at DSD128.' Question: What about Native DSD? 'DoP is a method to encapsulate DSD native signal in PCM stream. There is no conversion to PCM. It is true DSD signal, just that OSX thinks it is PCM. The DAC recognises it correctly and decodes the DSD signal. So DSD over PCM 1.0 is the correct setting'.
Is there is a DAC that will work with every Android phone (?) and if so why doesn't my iFi DAC work? Answer: This is not correct. The DAC in question maybe or is USB Audio Class 1.0 and does not support full HD audio (meaning they do not accept audio higher than 96kHz/24 bit and no DSD) so this is not to be confused with USB Audio Class 2.0. Therefore all this goes back to the USB Audio Player Pro website for a list of compatible devices with USB host mode. This is the most comprehensive list of compatible and non-compatible devices bar none. http://www.extreamsd.com/USBAudioRecorderPRO/
One customer reported the following on the latest iPhone X. 'USB accessories option, under Touch ID passcode options, turn this on.'
The following is from a customer: 'I went to the audio store today and tried a variety of DAC's with both official and unofficial USB male C to USB A female cables. Any time you plug in a USB male C to USB A female cable the iPad sets it as a headphone. In turn, it limits the output to 48kHz. So, until Apple fixes that no DAC will output above 48kHz, no DSD and no MQA. This is both the 11'and 12.9' 2018 iPad Pro. On the other hand, the new iPhone XR, XS and XS Max with a lightning to USB C cable work perfectly, and plays back DSD, MQA and 192khz+ files. If Apple switch the iPhone to USB C, this may result in exactly the same issue the iPad is having.'Past that, its probably worth trying Apple's official USB C camera connection kit:??https://www.apple.com/uk/shop/product/MJ1M2ZM/A/usb-c-to-usb-adapter?fnode=85&fs=fh=459d%2B4891
Apple are blocking off other vendor's cables via the use of codes/firmware so that you must purchase a genuine Apple product. Therefore we recommend only using Apple products. The following Youtube video of an Apple user that also explains this; https://www.youtube.com/watch?v=bNKK22PclgM Having said that, here are the required cable(s): http://store.apple.com/uk/product/MD821ZM/A/lightning-to-usb-camera-adapter For more information please read here: https://downloads.ifi-audio.com/home/computer-audio-tutorial/apple/
Please navigate to the Apple Logo > About this Mac > More Info > scroll down to System Report > Hardware > USB > iFi Device. Clicking the name of the iFi device should show you information including firmware version, VID and PID number.
Thunderbolt is a very niche interface, with a small installed base compared to USB and not widely adopted. To the best of our knowledge, there are no such 're-generator' devices for Thunderbolt. There are also very few Thunderbolt audio devices so we do not recommend one goes in this direction.
Thunderbolt combined PCIe and display port as well as DC power into a proprietary connector. Both these signals routinely employ re-driver/re-generate IC's at the circuit board level, so they definitely benefit from such devices.
The following is from a customer: 'Catalina blocks apps not digitally signed by Apple. You can get around it by going to System Preferences > Security & Privacy > General and enable any blocked app from Allow apps downloaded from pane at the bottom of the window'.
Our devices support DSD512 only via ASIO. If using WASAPI (Windows) or Core Audio (Mac) DSD256 is the maximum since DoP must be used.
For standard aptX latency is usually 150mS or less.
The xDSD is Bluetooth 3.0 compatible and supports: As far as aptX HD, LDAC, LDHC are concerned, they cannot be supported on the current Bluetooth system used by iFi, therefore, it is not possible to perform a firmware upgrade but generally speaking, aptX offers very high quality as is, for CD standard and streaming audio. With AAC and aptX we cover the largest installed base of existing users. If you're still having issues, why not contact our support team? We are available Monday to Friday to give you a helping hand.
The Bluetooth subsystem in iFi products is compatible with all current Bluetooth subsystems used in phones, computers and TV's etc. The Bluetooth version is generally immaterial for audio. What is relevant is the support of the A2DP device profile and for the desired codec's (e.g. aptX for laptops, desktops and android phones and AAC for iOS products from Apple). For example, having a Bluetooth V5.0 device not supporting A2DP audio at all will provide a much worse audio experience than having a Bluetooth V2.1 device that supports A2DP with AAC and aptX. So the key for audio on Bluetooth is A2DP profile and supported codecs, the actual version of Bluetooth is immaterial as all Bluetooth above 2.1 offer bandwidth etc. that is sufficient fo A2DP. A2DP Specification: https://www.dslreports.com/r0/download/2285126~a70eb148e16b921dc323dbb977d4b4b1/A2DP_SPEC.pdf https://www.dslreports.com/r0/download/2285126~a70eb148e16b921dc323dbb977d4b4b1/A2DP_SPEC.pdf https://www.dslreports.com 'The Advanced Audio Distribution Profile (A2DP) defines the protocols and procedures that realize distribution of audio content of high-quality in mono or stereo on ACL channels. The term'advanced audio', therefore, should be distinguished from'Bluetooth audio', which indicates distribution of narrowband voice on SCO channels as defined in Chapter 12 of Bluetooth Baseband specification. A typical usage case is the streaming of music content from a stereo music player to headphones or speakers. The audio data is compressed in a proper format for efficient use of the limited bandwidth. Surround sound distribution is not included in the scope of this profile.'
Although it may appear the LEDs are lit up, the LED's are actually'off', but there is a tiny amount of current flowing making the LEDs appear dim. The battery indicator should light up normally as per the manual.
Switching between settings has been engineered to ensure sonic transparency with advanced trench technology MOSFET is used as a muting switch. This FET-based switching is handled by a micro-controller, which only 'wakes up' when the user changes a setting, thus eradicating any sonically deleterious interference.
'Negative feedback' is used in amplifier circuits to compare the output signal with the input signal and correct errors, in order to control gain and reduce distortion. For sound quality, this is positive; BUT commonly applied, one-size-fits-all 'global negative feedback' can also highlight different problems at the same time as solving others ' corruption of the error signal, phase shifts, group delay and so on can all have a negative impact on sound quality. We recognised that different parts of a circuit benefit from specifically optimised feedback loops and have developed a negative feedback system that is much more accurate than the usual approach. This incorporates multiple feedback paths instead of a global loop, each path optimised for a particular function and working synergistically with the others to deliver optimal overall performance.
Balanced, differential analogue circuit design has long been championed for its ability to reduce noise and cross-talk within the signal path by fully separating the left and right channels. PureWave??is a new, balanced, symmetrical dual-mono topology with short, direct signal paths. The name refers to the sonic purity it achieves thanks to exceptional linearity and infinitesimally low levels of noise and distortion.
Linux does require DoP (DSD over PCM) as it has no ASIO driver support and DSD 512 only works on USB via ASIO not via other systems (Windows Audio Session / Core Audio / ALSA/PulseAudio). Currently, there is no ASIO support on MAC and we supply an ASIO driver that supports DSD512 for Windows.
In Linux, USB Audio Class 2.0 compliance are generally built-in. If not, they need to be added, however, we do not support Linux or provide drivers for it. You may need to acquire this from another source. This is because there are many different devices and lots of variables/variations. Please see the relevant standard documents at USB.ORG:??http://www.usb.org/developers/docs/devclass_docs/ (for 3rd parties like other manufacturers) A specific vendor or even individual may pick the bits that suit their purpose and formulate their own operating system (generally called a Distro aka 'Distribution'). What is included in this specific Linux Distro from the available repository of building blocks is down to the company or individual that assembles this. At the time of writing for this note??distrowatch.com??was tracking 281 different major active Linux Distro's: http://distrowatch.com/dwres.php?resource=popularity With so many variants (and smaller sub-variants such as audio targeted Distro's not even listed) it is impossible to provide any support for the use of iFi products under Linux or do effective testing. This best highlights the timeline for many, many versions of Linux. http://upload.wikimedia.org/wikipedia/commons/1/1b/Linux_Distribution_Timeline.svg
Our expertise lies in an in-depth understanding of the chipsets we utilize, surpassing that of other manufacturers. This unique insight allows us to tap into the latent capabilities of these chipsets effectively, exemplified by the Burr-Brown DSD1793's ability to achieve DSD512/PCM768. However, due to proprietary reasons, we cannot disclose the specifics of how we achieve this. It's essential to note that datasheets are typically created by the Marketing Department and may not fully represent the capabilities discovered by our Research and Development team. As evidence, we've previously unlocked features such as the DEM (Dynamic Element Matching circuit) in the TDA1541A of the CD-77, a capability absent from the official datasheet but successfully uncovered through our R&D efforts.
Using DoP (DSD over PCM) DSD is sent as a 16-bit chunk per PCM sample. So DSD (64) requires 4 PCM samples to hold 64 DSD bits and thus it needs to be sent as 176.4kHz PCM and thus DSD128 via DoP requires 352.8kHz PCM and DSD256 via DoP requires 705.6kHz PCM. Only via ASIO (native) is DSD sent at the DSD sample rate, though under the bonnet it is send in effect as 32 bit per PCM sample, so 705.6kHz allows 22.57M DSD sample rate (DSD512). The key difference between ASIO native and DoP is the way DSD signals are identified and how many bits are available per sample. ASIO has a mechanism separate from the data to signal DSD and uses 32 bit per PCM sample. For DoP the DSD marker is embedded in the PCM data stream and only 24 bit are used of which 8 bit are used as DSD marker.
If you mainly play CD's and/or CD derived MP3's at 44.1kHz, set to 44.1kHz/16 Bit and volume to max. This normally bypasses Windows internal processing, but not reliably (unfortunately). You must select a silent sound scheme (so that there is no system sound kicking in) and assign communication defaults to a different device (e.g. internal sound). In this case, as an example, there should only be the Spotify audio stream etc and Windows will normally bypass the mixer. However, as soon as two audio streams are mixed (e.g. system sounds and the main audio) the Windows mixer will kick in and stay engaged. It is possible to use third-party software (Audio Router, Equalify etc.) to operate so Spotify audio is sent to your DAC, while all system sounds remain on the built-in audio system, this case with a 44.1kHz/16 Bit default setting should be bit-perfect (fingers crossed). https://youtu.be/kekG3HBDNsM https://tall-paul.co.uk/2013/08/16/play-spotify-through-a-different-sound-device-windows/
Each steaming mode increases the USB buffer, which is set in X milliseconds. The ASIO buffer is listed in samples, in other words, how long it depends on the sample rate. Under all conditions, the ASIO Buffer must be larger than the USB buffer, or dropouts or other strange effects may happen. The USB streaming mode sets the following USB buffer length: Minimum Latency = 1 millisecond Low Latency = 2 milliseconds Standard = 4 milliseconds Relaxed = 8 milliseconds Reliable = 12 milliseconds Safe = 16 milliseconds Extra Safe = 32 milliseconds If using 768kHz sample rate (or DSD512 or DSD256 vis DoP) 8192 samples are around 10 milliseconds, so for correct operation at this sample rate the ASIO buffer setting just be the maximum 8192 and streaming mode must be relaxed (8mS). The downside is that such a setting will generate high latency at low sample rates, for example at 44.1kHz 8192 samples are 185 milliseconds, to which 8 milliseconds must be added. All of this is avoided if using the WASAPI audio subsystem. The latest 3.20 driver allows 'automatic' samples. The ASIO buffer is used to exchange sample data between the driver and an application (DAW). The ASIO buffer size is adjustable. However, because the ASIO buffer layer is driven by the USB streaming layer there is a dependency on the USB Streaming Mode setting. ASIO can work without dropouts only if the following condition is met: ASIO buffer size (in ms) >= USB streaming buffer depth (in ms) By convention, the ASIO buffer depth is specified in terms of samples which creates another dependency on the current sample rate. For example, if USB streaming mode is set to standard then the minimum ASIO buffer depth is 4 milliseconds which corresponds to 176 samples at 44100 Hz and 384 samples at 96000 Hz. Usually, an ASIO buffer size (in terms of samples) that is a power of two is preferred. In most DAWs sample processing is more efficient if such an 'even' number is chosen. So in the above example, we round up to the next power of two and end up with 256 samples at 44100 Hz and 512 samples at 96000 Hz. The driver internally does not perform buffer size checks and does not enforce an ASIO buffer size that still works with the current USB streaming mode setting according to the condition defined above. This logic is implemented in the control panel. The control panel allows the user to pick one of the power-of-two numbers between 64 and 4096 and then checks if this works with the current USB streaming mode setting (The driver provides an API function which performs this check internally). If the selected ASIO buffer is too small for the current setting then the control panel displays a warning message. In this case, the user should pick a larger ASIO buffer size value. As a consequence of the convention of specifying the ASIO buffer size in terms of samples, the user has to adjust the ASIO buffer size every time the sample rate in a DAW is changed.
Windows will only show sample rates up to 192kHz. We recommend that you use a suitable playback software (e.g. JRiver). Under no circumstances would we suggest to use the Windows audio subsystem to up-sample audio, the SRC (Sample Rate Conversion) build into windows is quite poor.
The number of buffers literally means the number of samples that are held in memory by the driver. Larger buffer numbers mean there is less risk of the driver running out of audio samples if the computer is busy with something else. Very high sample rates (e.g. 768kHz) require maximum buffers on all but the most powerful hardware. We are less sure about the streaming mode, however, it seems in part to relate to how much resources on the PC the driver uses. Extra safe usually is fine to use, it gives the driver the biggest 'footprint' and makes sure transmission is least likely to be interrupted by other processes.