Chord Mojo(1) DAC-amp ☆★►FAQ in 3rd post!◄★☆
Oct 14, 2015 at 8:56 AM Thread Starter Post #1 of 42,758

Mython

Headphoneus Supremus
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2023 note: Mojo 2 thread can be found HERE







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Mojo 1 users, please Click Here for Frequently Asked Questions and In-Depth Information










Chord Mojo awarded European Imaging and Sound Association's

'European USB DAC/HEADPHONE AMPLIFIER 2016-2017'!
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Chord Mojo awarded WHAT HI★FI? 'Product of the Year', 2015!

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Mojo has landed



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Chord Electronics, a world leader in digital audio, has launched Mojo.

A contraction of ‘Mobile Joy’, Mojo is a headphone amplifier and DAC (digital-to-analogue convertor) that empowers smartphones to deliver music content at up to studio-master-tape quality.



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Low-cost, widely available apps, such as Onkyo’s HF player (iOS and Android), now make high-resolution music files playback easy from all smartphones. Mojo connects to these devices digitally, processing the files using the most advanced conversion technology available, to deliver genuinely unrivaled sound quality to up to two pairs of headphones (You can use any pair of headphones with Mojo, from 4Ω to 800Ω).

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Mojo has three digital inputs; Micro USB, optical, and Co-ax, and has been designed to work with your iPhone, iPad, Android phone (USB OTG), Android tablet (USB OTG), Mac, PC, and Linux computer.

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Despite Mojos ultra compact form, Mojo takes just four hours to fully charge and can deliver up to eight hours continuous use. But, thanks to Mojos separate Micro USB charging port, you can play and charge at the same time. External power banks can be used to charge Mojo on the move so long as they have at least 1 amp output.




Mojo is capable of playing all of today’s music formats, including the very latest high-resolution standards. It can deliver breath-taking realism from any digital music file: PCM; WAV; AAC; AIFF; MP3 and FLAC. It is designed to work with all smartphones and music players, and covers specialist high-resolution formats such as DoP DSD files: DSD 64; DSD 128 and DSD 256. Mojos three high-resolution digital input options comprise optical (to 192kHz), plus MicroUSB and RCA (mini-jack) which operate at up to an incredible 768kHz.

Mojo is entirely designed & manufactured in England.

UK RRP: £399
US RRP: $599





Technical specification

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Mojo uses a Xilinx Artix 7 FPGA

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Last edited:
Oct 14, 2015 at 8:56 AM Post #2 of 42,758

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Head-fi Mojo FAQ (started by 'Currawong') (NB: this should be considered supplementary to the comprehensive information, below)​
 
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  Here are my slides from the Shard presentation:
 

 

 

 

 

 

 

 

 
 

 
 
 
Quote:
  Its a brain issue, and is (mostly) down to two technical problems - one being noise floor modulation, one being timing uncertainty. With timing uncertainty, when the sampled digital data is converted back to a continuous signal, the DAC creates timing errors. These timing errors then interfere with the brains ability to actual perceive the starting and stopping of notes - and when the brain can't easily recognise something, it has to work harder to make sense of what is going on. Its a bit like one being in a party trying to understand somebody speaking with a lot of noise - your brain has to work harder to understand the voice, and its tiring. The noise floor modulation problem, means that the brain has greater difficulty separating sounds out into individual entities. What people forget, as we take hearing for granted, is that the brain is processing the data from the ears, and separating things out into individual entities, and also putting a placement tag onto that entity. Noise floor modulation makes it more difficult for the brain to separate things out into individual entities, so the brain has to work harder to make sense of the music. And when it has to work harder, you get listening fatigue.
 
Now the timing issue is a unique problem with digital audio, and noise floor modulation is about ten times a larger problem than with amplifiers, so you can see why listening fatigue is a particular problem with digital.
 
Rob
 
Quote:
 
Is it my brain. Or is it that Mojo just gets better and better the longer I use it? This little gadget is really stunning.

I had same experience with Hugo. It just seemed to get better and better, and took 9 months before the feeling of improvements stopped. Funny thing was it was not break in as new Hugo sounded the same. My assumption was my brain breaking in to the way that Hugo recreated transients which was quite different to any other dac before.
I expect Mojo to be the same.
Rob
 
Quote:
  Electrolytic capacitors take time to break in - leakage current takes 3 months to minimize and so does ESR (equivalent series resistance). If you use them in the audio path (I do not) then bass distortion gets lower with time. It is possible to reduce break-in time, and I do this.
 
With Hugo I kept on getting the feeling that SQ was getting better and better - even nine months on - but when given brand new product from Chord, once warmed up, they sounded the same. So it was not the hardware, and either I was deluding myself, or my brain was un-learning digital music. Now Mojo/Hugo/Dave do things in the time domain that no other DAC's do, so its easier for the brain to make sense of the music as timing of transients has much less uncertainty. Certainly the brain does get used to a particular sound, and creates processing short cuts that allows better understanding of the sound, so its not a great leap to state the possibility that our brain's unlearn digital sound as after all, we are surrounded by it.
 
I can say that since Hugo I can no longer tolerate listening to music using conventional DAC's.
 
Rob  

 
Quote:
  The brain breaking in problem I think is actually about us dealing with conventional digital audio - we listen to digital music all the time - TV, Hi-Fi and portable gear and actually listen to un-sampled music probably less than 1% of the time - so our poor brains is saturated by having to deal with the timing problems of sampled digital music - and I guess it has created coping strategies to deal with uncertainty in the timing of transients. Then along comes something different, with the timing uncertainty removed - so the brain has to unlearn the coping mechanisms.
 
Now I am of course only guessing here, but it was very odd when I first heard Hugo - and that took 9 months for me to get used to the sound - I had this constant feeling that SQ was getting better - it was not Hugo, as new units sounded the same as my old unit.
 
Having said all that, the first ten seconds of listening to Hugo I knew immediately that something very strange was going on, as the sound was very different to what I was used to, so people should hear the difference that Mojo makes very quickly. But I have the benefit of being an experienced and sensitive listener.
 
The really curious thing about all this is that the actual timing differences in terms of error is very small - the ear/brain is a remarkably sensitive system, and science has little understanding about things we take for granted, such as out perception of sounds. How does the brain separate sounds out into discrete entities and put an extremely accurate placement tag on it? There is some amazing processing going on for which we have no understanding.
 
Rob  

 

Also see John Franks' remarks: https://youtu.be/DTWcKLI0g7c?t=41m7s
 
 
 
  I got my Mojo yesterday, and have been comparing to my iDSD micro. I hear what people refer as "musical" and "emotional," but I want to understand why. The iDSD sound more flat, more monotone almost compare to the Mojo. Is the Mojo more true to the original sound? I'm not sure, but it sure sounds more fun. It's almost like describing a picture, the Mojo has more contrast and vibrancy, where as the iDSD has a more flatter, but possibly more true color.
 
Then there's the matter of warmth. The iDSD is known for its warmth, yet that warmth has a bit of way veiling the sound. Mojo also has a hint of warmth, but that warmth is applied without extra veil. I'm in no way dismissing the iDSD, as it is absolutely superb, and perhaps more neutral. But the Mojo just sounds more exciting than the iDSD. I'm comparing using both my HD800 and the Mainline as amp, as well as my SE846 directly plugged into the Mojo/iDSD unit.
 
One thing for sure though, the Mojo is much more compact and portable than the iDSD. Too bad I'm not using them as portables and strictly as desktop DAC's. The iDSD is more desktop friendly and allows easier charging while being used. I haven't tried both charging the Mojo while it's being used, but I read that it uses battery faster than being charged, being in the desktop mode. Since it has more settings, the iDSD is also more compatible with a wider range of headgear, be it IEM or high impedance full sized headphones. I just love the build on the Mojo though, it's truly a work of art.

 
 
 
Relating sound quality to technical performance is very complex, and I will try to explain, but I could talk for days about it and completely confuse everybody. But here is a quick answer to your questions.
 
 
   I hear what people refer as "musical" and "emotional," but I want to understand why.

 
Musicality and emotional is complex, but in a nutshell its about removing distortions that interfere with the brains ability to understand the music. Conventional DAC's have a number of distortions that make it much harder for the brain to perceive the sound. Now we underestimate what the brain does with hearing, and simply make the assumption that the ears convert sounds into nerve impulses, and that's that job done, the brain simply access's the nerve signals. But that's not what happens - audible reality is an illusion created by the brain, and a considerable amount of brain processing is employed to create that illusion. So for example, you listen to a guitar and a singer for example. The data the ears feed the brain is a jumbled up mess of information, and the brain separates this mess of data into two distinct entities - the guitarist and the singer, and you perceive this as two separate entities. Not only that, but the brain very cleverly calculates where in space those entities are, and it does this from subtle timing, amplitude and resonance cues from both ears. But this requires considerable calculation. Moreover, small and subtle distortions (by saying distortion I mean anything that changes the original signal in any non linear way) interferes with the brains ability to separate sounds out into distinct entities, and interferes with the brains ability to place entities in space. This has two consequences for being able to enjoy music - firstly the brain is struggling to process the data, so has to work harder - which means you get listening fatigue, and so you can't enjoy the music. Secondly, being able to enjoy the music means being able to perceive what is going on - and there are many distortions that disable the brains ability to perceive the music. This is where it gets complex, as there are a myriad of different distortions that upset the brains processing. That's why Mojo has the WTA processing, why it filters and over-samples at 2048 times, why its got noise shapers that are a thousand times more resolving than conventional noise shapers - I could go on.
 
 
 
  The iDSD sound more flat, more monotone almost compare to the Mojo.

 
The perception of depth information is down to very small amplitude differences of small signals. Now the brain calculates depth from a number of different cues, but most of it comes from the reverberant sound from the acoustic the recording was made in (or depth is added by adding artificial reverb). Now reverb is very small signals, and the amplitude accuracy of these small signals is crucial for the brain's ability to calculate depth. Now there is something very strange about depth perception - and that is the brain needs these small signals to have perfect amplitude linearity. If a small signal is slightly larger or slightly smaller than it should be, then the brain gets confused and can't calculate the depth properly, and things then sound flat. But the amazing thing is, there appears no limit to how accurate these small signals need to be in order for the brain to not truncate or flatten depth. In order to accurately reproduce depth you need extreme small signal linearity. You can't do this with R2R DAC's, as the resistors can't be matched. With DSD or delta sigma (Mojo is delta sigma too) the problem is now how well the noise shaper functions. As a signal gets closer to the noise shaper noise floor, the levels get smaller, as a signal that is smaller than the resolution limit of the noise shaper is truncated. To overcome this you need to have very high resolution outputs, with a noise shaper that has very high resolution - in Mojo's case, the noise shaper has a thousand times more resolving power than conventional high end noise shapers, and ten thousand times more resolution than DSD 64. But there is another source of error that can upset sound-stage depth and this is digital noise adding to the analogue signal. This applies to all DAC's, and is a big problem with chip DAC's, as there always exists a path from the digital noisy part to the analogue part, and this noise corruption will degrade the small signal non-linearity. But with Mojo the actual analogue parts are discrete, so its possible to eliminate digital noise from corrupting the signal. There is another mechanism for depth to be truncated, and this is with metal to metal interfaces. When you have a soldered joint, or any metal to metal interface, oxides and impurities concentrate at the interface. This oxide barrier is non-linear in that the resistance to small signals is larger than with big signals - so again we have small signals being attenuated. To reduce this problem you can only do this by reducing the number of passive components in the signal path. Conventional DAC's (delta sigma and R2R) have very complex analogue components, due to the need to convert from differential to single ended and to filter the high amounts of RF that comes out of a conventional DAC. With pulse array (my DAC technology within Mojo) this is not an issue as I can get single ended to work, and it runs at 104MHz, so little analogue filtering is required.
 
 
 
  The iDSD sound more flat, more monotone almost compare to the Mojo.

 
I think here you are referring to timbre - the tonal colour of the instrument. Now timbre is an issue with timing reconstruction, as the brain uses transient information to infer the timbre of an instrument. Now conventional digital has uncertainty in the timing of transients (does a signal cross through zero just after a sample, or in the middle or close to the end of a sample?) and the only way of recovering the timing information perfectly is to use an infinite amount of processing on the interpolation filter. With the use of the WTA filter, which has been optimised to recover timing, and 500 times more processing than conventional DAC's, I can reduce the timing uncertainty - which results in much better timbre variation, so things don't sound monotone. There is another aspect in that noise floor modulation also affects timbre reproduction, but this is answered in your next question.
 

 
  Then there's the matter of warmth. The iDSD is known for its warmth, yet that warmth has a bit of way veiling the sound. Mojo also has a hint of warmth, but that warmth is applied without extra veil. I'm in no way dismissing the iDSD, as it is absolutely superb, and perhaps more neutral.

 
Warmth or smoothness can be artificially created - for example with a dollop of 2nd harmonic. Mojo has very low levels of distortion, so its warmth is not down to doing this or other things. The key to true refinement with DAC's is noise floor modulation. This is where the noise pumps up and down with the signal, and all other non Chord DAC's have large amounts of noise floor modulation. Now noise floor modulation is a scary issue with DAC's, and there are countless ways that a DAC can suffer. Mojo, on the other hand has zero measurable noise floor modulation - the noise floor is at -170dB and it maintains this whether its output is 2.5v or zero, the noise is completely static. Now the issue of neutrality is a very complex thing, as increasing transparency will make it brighter and sharper, and increasing refinement will make it smoother and darker, and its possible to use distortion to create the impression of warmth or brightness. To be honest, I (or anybody else for that matter) do not know what the tonal balance of a perfect (and hence neutral) DAC is. And neutral cam mean different things to different people and with different gear!
 
Mojo's musical performance is down to lots of technical things - way too complex to talk in detail with - but there are solid reasons why you hear what you hear, and why other DAC's can't do this.
 
Rob


 
 
Quote:
  There has been some recent discussion about digital filters, in particular closed form mathematics. There is a lot of confusion about what is actually happening, and this is not surprising - filter design is complex, and people talk about things that they have little real understanding.
 
Indeed, the more time and work I spend in audio, the more I realise how much more there is to know - we are all scratching at the surface, so some humility is needed. "You know nothing Jon Snow" is my favourite quote from Game of Thrones, and I often bear it in mind when thinking about audio, and how to relate something I hear with theory.
 
Now there are two things that are talked about closed form filter design - one being that the the filter coefficients (these are fixed at the design of the filter) uses a closed form algorithm which just means that it is a formula to calculate the numbers. The second issue is that the initial filter samples are preserved.
 
Now most FIR filter algorithms are closed form. The exception, as pointed out by a poster earlier is the Parks–McClellan which uses the Remez algorithm to iteratively calculate the optimal solution for the coefficient calculation. It is not a closed form calculation, as it cleverly runs backwards and forwards until it converges onto the desired result. Now is a closed form a good or a bad idea? Frankly, it does not matter how the coefficients are calculated, its what those coefficients are, and what they sound like that is important. Now I don't like the Parks-McClellan algorithm, as it does not maximise rejection at the points where there is the most out of band energy which is at FS multiples. And its not very good at recovering timing information for the intermediate samples you are trying to create. But this is not closed form or iterative process that is important here. Now the WTA algorithm is closed form, you can calculate the ideal coefficients to as much accuracy as you like with one fixed equation. But whether it is closed form or not is just unimportant.
 
The second issue is exactly maintaining the original samples. Now the vast majority of FIR filters for audio are known as half band filters, and to create a 8 times oversampled filter you use a cascade of 3 half band filters. These are guaranteed by design to give the original data, and they are used because they are computationally efficient, as half the calculations are zero - you simply return the original sample, no maths. Most are designed with Parks-McClellan, so the issue of closed form has actually nothing to do with retaining the original sample data.
 
So maintaining the original sample data is a red-herring as regards closed form. But is keeping the original data actually a good idea? It sounds like a great idea, why mess with the actual data?
 
When I was developing the WTA algorithm in the late 1990's I hit a stumbling block. I had designed a very long tap length half band filter - so it was 2048 taps, half being zero, so it returned the original sample perfectly. It sounded very much better than the filters I had before, but I knew that timing recovery and transient accuracy was a problem. I could see also that aliasing issues from the half band filter would degrade transient accuracy, so I needed to remove these measurable aliasing problems. But that would mean the original data would get changed, and I did not like that.
 
One trap that designers and audiophiles fall into is to think doing XYZ is wrong and that it must sound better because of this particular idea. That is a very easy trap to fall into - or even think some idea must sound better, then listening to it, then convincing yourself that this soft muddled sound is actually better (or this bright hard sound is more transparency and at last I can hear how bad recordings actually are). In other words your thinking is convincing yourself that something is better (of course your lizard brain is not fooled and you end up listening to less music and enjoying it less). I too was stuck in the trap that the best thing to do was to keep the original data. But at the end of the day, you got to try it, do careful listening tests, and run by the evidence, not what you think may sound good or con yourself into thinking something is better. So eventually I tried eliminating the reconstruction aliasing, and boy did this make a big improvement - even though the samples were not being preserved - bass was much deeper, sound-stage much more accurate, and the flow and timing much more natural. 
 
So some humility is called for, nobody has a perfect understanding of anything, and thinking something must sound better is extremely dangerous. Do the work, listen carefully and neutrally, and base everything on the evidence, not on attractive ideas.
 
Rob

 
  PC's are very restricted in what they can do for real time signals. You simply can't replicate the processing that Dave does in a PC - simply because PC processors are sequential serial devices with a very limited number of cores. When you are doing a doing a FIR filter (a tap) you need to read from memory the audio data; read from memory the coefficient data; multiply the numbers together;then read the accumulated data and add that to the previous multiplication; then save the result. Lots of things to do in sequence. With an FPGA you can do all of these things in parallel at once, so a single FIR tap can be accomplished within a single clock cycle (obviously pipelined) - you are not forced to do things in sequence.
 
With Dave I have 166 dsp cores running, plus FPGA fabric to do a considerable amount of further processing. You simply can't do that in a PC. To give you another example - converting DSD into DoP. You need a quad core processor to do this manipulation in real time - otherwise you get drop-outs - but in a FPGA I could do this simple operation thousands of times over, and at much faster rates than DSD256.
 
What some people do not understand is how capable FPGA's are and how widespread they are used - the backbone to the internet? FPGA's. Search engines? FPGA's. Why? because an FPGA is fantastic at doing fixed real time processing - it takes small die area, and can do complex operations with very low power. Mojo for example has 44 dsp cores, uses sophisticated filtering to 104 MHz, and noise shapes at this rate - but does all this whilst consuming only 0.45 W. There is no way any PC consuming huge amounts of power can do this.
 
Intel last year acquired Altera (an FPGA company) for $16.7 billion because they understand that the future of processing is with FPGA's
 
A second issue is not what you can do but how you can do it - it is not just about raw power, but how the filter algorithm is designed. I have put many thousands of hours and over twenty years improving and understanding how to make a transparent interpolation filter; and I am still learning things today.
 
And a third point is that a DAC is not simply a data processing machine but it has got crucial analogue parts too. If I dropped the WTA requirement, I would still need the same FPGA in order to do the noise shaping and other functions.
 
Rob

 
 
 
@robwatts @mojo ideas

How about an impedance module that allowed us to adjust output impedance until we perfectly matched mojo to our ciems/headphones?

The technically perfect impedance is zero, and that's why I worked so hard to get it as low as 0.075 ohms with Mojo.
 
The reasons going for as close as zero are:
 
1. Frequency response. The impedance of the headphone varies with frequency, and so by having a high output impedance will cause frequency response variations. Zero impedance eliminates this problem.
 
2. Distortion. The impedance of a headphone varies with level, and having a higher output impedance will increase the total distortion - given that Mojo distortion is so low, this is actually quite a significant an effect. Again, zero impedance eliminates this problem.
 
3. Damping factor - probably the most important reason. A drive unit is a resonant system - that is a mass on a spring - that is damped mechanically and electrically. Electrical damping is due to the headphone creating a current due to the motion of the driver in the magnetic field - and how well this is controlled depends on the electrical impedance the driver sees - in our case, the cable impedance and Mojo's impedance. Again, zero impedance gives the best damping, with an infinite damping factor.
 
I did some listening tests many years ago with loudspeakers and damping factor and found that it made a massive difference to the sound. Damping of 10 gave a very soft, big fat bass - but everything sounding one note in the bass - simply because the loudspeaker was doing its own thing at the resonant frequency. Going from 10 to 100 gave a tighter bass, with much better pitch reproduction - you could follow the bass line much more easily. Above 100 to 1000 it sounded tighter - no big change in pitch (being able to follow the bass tune) but the perceived tempo of the music became faster as transients are much better controlled. Going above 1000 gave a small improvement in how tight it sounded.
 
Rob

 
 
 
 
What does *laid back* mean in sound? I never understood. Is mojo considered laid back
?

 
Laid back or relaxed tends to suggest less dynamic punch and a less forward sound. Less in your face is another way to put it. This can also mean less fatiguing.

 
OK I appreciate it, it makes sense, I like my sound a little punchy and in your face, but also not fatiguing haha, mojo definitely *warmed* up my sound and I enjoy that, because it still stays very clear and detailed

I often think about this issue as yin-yang (dark-bright), and a good product has this in balance - but what the correct balance is does depend somewhat on taste!
 
So yin - dark - is in technical terms, happens with zero noise floor modulation. Conventional DAC's have enormous levels of noise floor modulation. This means noise (bright hiss) pumps up and down with the music signal, and the brain can't separate a dark sounding instrument from the noise floor modulation - so smooth sounding instruments become bright. With Chord DAC's, including Mojo, there is no measurable noise floor modulation, so it innately sounds smooth and warm.
 
But its possible to artificially give the appearance of more yin by contouring the sound. For example, add a lot of second harmonic distortion, and it sounds thicker and darker - but its an illusion, as everything sounds soft. You can also add LF errors too, to give the impression of more weight to the sound - adding electrolytic caps, or letting the ref circuitry amplitude modulate the output from the signal envelope. Indeed, a lot of designers rely on this, as they do not have the abilities (stuck with using chip DAC's) to solve noise floor modulation, so have to use tricks to balance the sound.
 
On the yang side, natural brightness comes from two sides. First is transparency, and this resolves into detail resolution, and this is about how accurate the DAC/amp can resolve very small signals accurately. With my work on the reference DAC Dave, I discovered that there is no limit to how accurate the small signal needs to be - the smallest possible amplitude error is very audible, particularly in terms of sound-stage depth. Transparency is a complex issue, but comes down to two main issues - simplicity of the analogue section (each component degrades small signal linearity) and the performance of the noise shaper (before anybody says ladder DAC's these are awful for small signal linearity). Now Mojo has an extremely simple output stage - only one active stage and two resistors and two capacitors in the direct signal path, and this is done for transparency. On the noise shaper, it has 1000 times more resolution than conventional noise shapers, as the noise shaper runs at 104 MHz, not the usual 6 MHz of the best chip DAC's.
 
The second part of yang is timing. Now digital audio is sampled data, but the original signal in the ADC is a continuous signal, and the job of the DAC is to convert the sampled signal into a continuous analogue signal with the timing of the original signal in the ADC perfectly preserved. Now I talk a lot about reconstituting timing, and have had requests to show the problem. So here is a simple illustration of the problem:
 

 

 

 
Now this is a bit of a simplification - the burst signal is not bandwidth limited, but it serves to illustrate the problem of timing inaccuracies. Now how do these timing errors sound like? When the brain comes across timing errors, it can't deal with it - it can't make sense of the music. And when the brain can't process the signal, you then can't hear the transients. It is a bit like putting a picture out of focus, blurring the edges. What this does audibly is to make transients sound soft, and when one improves timing accuracies then the brain can perceive the starting and stopping of notes accurately - so things sound sharp and fast - more yang. Now what is curious about timing errors, is that there again is almost no limit to how small they need to be - before Dave, I used to think in terms of uS errors, now its definitely nS as being important - extremely small timing errors have a noticeable subjective musical impact. 
 
 
Also it is very possible to use distortions to give impressions of good sound - use slew related noise floor modulation and you get the impression of good timing resolution - but its entirely false. The problem with using distortions like this, although it can sound superficially impressive - is that everything always sound the same. But the major problem with this approach is simply listening fatigue - I can listen to Mojo for 10 hours and still want more. It also illustrates the design nightmare of listening tests - is the sound quality "improvement" real or just more distortion or aberration? You have to be extremely careful on how one assesses sound quality.  
 
So to conclude - Mojo can sound both rich & dark (immeasurable noise floor modulation) & very fast & dynamic (much lower timing errors) all at the same time. That's why we get so many different reactions to the sound of Mojo - some saying its rich & smooth, some saying its fast and dynamic - & the truth is both observations are correct.
 
Rob

 
 
 
Quote:
 
  Digital transmission is based on SPDIF standard which transmits data and clock information as an encoded signal usually using PCM, that information is decoded on the Mojo into data and clock signal so it's important that the encoded information be jittered free and not degraded over short distance.
 
The USB transmission on the other end is a device to device transmission mechanism using an encoding scheme and handshaking mechanism, it is usually stream based so more tolerant to poorer wire as frames are transmitted and decoded from the source to the target device. The target device will reconstruct the data and clock signal from the frame and then feed it to the DAC to be analog reconstructed and eventually band pass filtered to remove any residual high and low frequency signals out of the audio band.I still think you need to keep the USB cable short but it is more tolerant of longer lengths up to a limit.
 
To make a story short, the short USB cable is fine but an analog cable used as a digital one is just a bad idea. Again, that's just my opinion.


Just to clarify:
 
1. SPDIF decoding is all digital within the FPGA. The FPGA uses a digital phase lock loop (DPLL) and a tiny buffer. This re-clocks the data and eliminates the incoming jitter from the source. This system took 6 years to perfect, and means that the sound quality defects from source jitter is eliminated. How do I know that? Measurements - 2 uS of jitter has no affect whatsoever on measurements (and I can resolve noise floor at -180dB with my APX555) and sound quality tests against RAM buffer systems revealed no significant difference. You can (almost) use a piece of damp string and the source jitter will be eliminated.
 
2. USB is isochronous asynchronous. This means that the FPGA supplies the timing to the source, and incoming USB data is re clocked from the low jitter master clock. So again source jitter is eliminated.
 
So does this mean that any digital cable will do?
 
Sadly no. Mojo is a DAC, that means its an analogue component, and all analogue components are sensitive to RF noise and signal correlated in-band noise, so the RF character of the electrical cables can have an influence. What happens is random RF noise gets into the analogue electronics, creating intermodulation distortion with the wanted audio signal. The result of this is noise floor modulation. Now the brain is incredibly sensitive to noise floor modulation, and perceives this has a hardness to the sound - easily confused as better detail resolution as it sounds brighter. Reduce RF noise, and it will sound darker and smoother. The second source is distorted in band noise, and this mixes with the wanted signal (crosstalk source) and subtly alters the levels of small signals - this in turn degrades the perception of sound stage depth. This is another source of error for which the brain is astonishingly sensitive too. The distorted in band noise comes from the DAP, phone or PC internal electronics processing the digital data, with the maximum noise coming as the signal crosses through zero - all digital data going from all zeroes to all ones. Fortunately mobile electronics are power frugal and create less RF and signal correlated noise than PC's. Note that optical connection does not have any of these problems, and is my preferred connection. 
 
Does this mean that high end cables are better? Sadly not necessarily. What one needs is good RF characteristics, and some expensive cables are RF poor. Also note that if it sounds brighter its worse, as noise floor modulation is spicing up the sound (its the MSG of sound). So be careful when listening and if its brighter its superficially more impressive but in the long term musically worse. At the end of the day, its musicality only that counts, not how impressive it sounds.         
 
Rob
  The reasons why sources and digital interconnects sound different are well understood - see some of my posts. In a nutshell it is not jitter (all my DACs are completely immune to source jitter) but down to RF noise and distorted currents from the source flowing into the DAC's ground plane. The RF noise inter-modulates with the analogue electronics, creating random noise as a by product, which creates noise floor modulation, and that makes it sound brighter or harder. The correlated or distorted currents very subtly add or subtract to small signals, thus changing the fundamental linearity, which in turn mucks up depth perception.
 
But I also agree in that lots of people hear changes that are not there - I for one have never heard any difference with optical cables (assuming all are bit perfect) with my DAC's, but lots of folks claim big differences. Placebo, or listening with your wallet, plays a part here. Then there are cases of people preferring more distortion... Listening tests must be done in a very controlled and careful fashion, particularly if you are trying to design and develop things.
 
Rob

 
 
At the risk of being flamed, I don't see how the composition of the USB cable wire can add warmth to digital data from whatever device is being used as a transport

I understand those concerns too - after all the data is the same. But there are solid scientific reasons why they can make a difference.

 
In the 1980's, people started talking about mains cables making a difference to the sound quality - and I didn't believe it either - particularly as my pre-amp had 300 dB of PSU rejection in the power supply. But I did a listening test, and yes I could hear a difference. Frankly I still could not believe the evidence of my own ears, so did a blind listening test with my girl friend. She reported exactly the same observation - mains cables did make a difference to SQ.
 
To cut a long story short, I proved the problem was down to RF noise. RF noise inter-modulates with the wanted audio signal within the analogue electronics, and if the RF noise is random, then the distortion is random too and you get a increase in noise floor with signal. This increase in noise floor is noise floor modulation, and the brain is very sensitive to it; you can perceive tiny amounts of noise floor modulation as a brightening or hardening of the sound. By tiny I mean the noise floor modulation needs to be well below -200 dB, so the brain is very sensitive to it. With the right test equipment, you (APX5555 is only test equipment that has no innate noise floor modulation) can easily measure the effect.
 

The RF characteristics of the cable can change the RF noise that gets injected into Mojo's ground plane, and this is the mechanism for changes in smoothness. You may say why can't you make it insensitive to it; well I go to silly lengths to RF filter and decouple, and use dual solid ground planes on the PCB, but you can't remove the problem. For Dave, Hugo TT and 2 Qute I have galvanic isolation, and this eliminates the problem (along with other SQ problems such as sound-stage depth). But I can't do this with portable devices, as it draws power from the 'phone. That said it's less of an issue with portable electronics as they are less power hungry and create less noise.

 

So what are the best USB cables? Firstly, be careful. A lot of audiophile USB cables actually increase RF noise and make it sound brighter, and superficially impressive - but this is just distortion brightening things up. Go for USB cables that have ferrites in the cable is a good idea - it may also solve any RF issues from the mobile that you may have too.

 

Rob

 
 
Quote:
 
Clock jitter -- What is clock jitter? The reason I ask is that in considering different DAPs to use as transports to output a digital signal to the Mojo, I've seen some varying specs for clock jitter on different DAPs, as follows:

AK100 90 ps (pico seconds)
AK120 50 "
AK240 50 "

Question: Does clock jitter degrade the digital signal before it's send out from the DAP? Or are they referring to clock jitter of the internal dac, in which case clock jitter doesn't matter since the signal never reaches the dac (it's been output beforehand)?

If clock jitter degrades the signal before it's sent out, then it appears that the AK100 is not as good a transport as the other two. But would the difference be discernible?

Thx

 
Clock jitter is timing uncertainty (or inaccuracy) on the main clock that is feeding the digital outputs. Its often expressed as cycle to cycle jitter as an RMS figure, but can be total jitter which includes low frequency jitter too. Total jitter is the most important specification. If you want here is a good definition:
 
https://en.wikipedia.org/wiki/Jitter
 
As you can see, the jitter subject can get complicated and its often abused by marketing...
 
But with all of my DAC's you do not need to worry at all about source jitter, so all of the above AK numbers are fine. So long as its below 2uS (that is 2,000,000 pS) you are OK, and nobody has jitter that bad!
 
Rob
 
Quote:
  .... Mojo has the same USB as Dave - but with Dave it is galvanically isolated, so RF noise and signal correlated currents from the source can't upset Dave at all. I can't do this with Mojo as it draws too much power from the device connected to the USB - but the upside is that mobile sources create much less noise as they are power efficient and there is no ground loops either due to battery operation.
 
So Mojo like Dave has solved the jitter problem via USB, as the timing for the data comes from Mojo not the source.
 
Rob

 
Quote:
  The reasons why sources and digital interconnects sound different are well understood - see some of my posts. In a nutshell it is not jitter (all my DACs are completely immune to source jitter) but down to RF noise and distorted currents from the source flowing into the DAC's ground plane. The RF noise inter-modulates with the analogue electronics, creating random noise as a by product, which creates noise floor modulation, and that makes it sound brighter or harder. The correlated or distorted currents very subtly add or subtract to small signals, thus changing the fundamental linearity, which in turn mucks up depth perception.
 
But I also agree in that lots of people hear changes that are not there - I for one have never heard any difference with optical cables (assuming all are bit perfect) with my DAC's, but lots of folks claim big differences. Placebo, or listening with your wallet, plays a part here. Then there are cases of people preferring more distortion... Listening tests must be done in a very controlled and careful fashion, particularly if you are trying to design and develop things.
 
Rob

 
Quote:
  There are two problems that USB has against toslink - and one benefit. The benefit is that timing comes from Mojo - but with toslink the incoming data has to be re-timed via the digital phase lock loop (DPLL) and this is not quite as good - but you will only hear the difference via a careful AB test, so it's in practice insignificant.
 
The downside with USB is the common ground connection. This will mean RF noise will get into Mojo, making noise floor modulation worse. Now I go to very careful lengths to remove this problem by using lots of RF filtering, and double ground planes on the PCB, but even minute amounts of RF is significant. The other problem is down to the way that digital code works - which is in twos complement. So zero is in 24 bits binary is 0000_0000_0000_0000_0000_0000. If the signal goes slightly positive then we get just one bit changing to: 0000_0000_0000_0000_0000_0001. But if it goes 1 bit negative all the bits change to:  1111_1111_1111_1111_1111_1111. Now the problem with this is that when a bit changes, more power is needed, and this injects current into the ground of the PC - and the ground will get noisier. Unfortunately the noise is worst for small signals. Now the problem with this is that it then couples through to Mojo's ground plane, and the distorted signal currents will add or subtract to small signals - thus changing the small signal linearity. This in turn degrades the ability of the brain to re-create depth information, and so we hear it in terms of depth being flattened. What is really weird about depth perception is that there seems to be no limit to how accurate it needs to be, so the smallest error is significant.
 
So with toslink we do not get these problems as there is no common ground - so no RF noise, no distorted signals on the ground, and it will sound smoother with better depth against a noisy PC. But the problem can be almost eliminated by using a power efficient USB source that is battery powered - such as a mobile phone. But with noisy PC's the only way of solving it is to use galvanic isolation on the USB - but this draws power from the source, and we can't do that with mobile devices. All of Chord's desktop DAC's have galvanic isolation on the USB, and then you can't hear whether its a noisy PC or a mobile phone. In this case, USB sounds slightly better than optical, because we have the (tiny) timing benefits of USB.
 
I hope that explains - its a complex subject.
 
Rob

 
 

 

 
 
  .... the volume control is in the central WTA filter core, and has an internal accuracy of 51 bits. But it then gets passed to the cross-feed dsp, then on to the 3 stage interpolation filters to take it to 2048FS, then into the OP noise shapers. So the 51 bits has to be truncated. But since the signal is at 16FS, the truncation is done via noise shaping and dithering. This means that the signal is not lost, but perfectly preserved, as this process adds zero distortion - just a fixed noise at -180dB. This has been verified with Verilog simulation.  
 
Rob

 
 
  Just to make it 100% clear - the USB input will measure absolutely identically to the coax or optical inputs if the USB data is bit perfect.
 
I have set up my APX555 so that it uses the USB via ASIO drivers, and I get exactly the same measurements on all inputs - 125 dB DR, THD and noise of 0.00017% 3v 1k 300 ohms. I have done careful jitter analysis, FFT analysis down to Mojo's -175dB noise floor, and can measure no difference whatsoever on all inputs (with the APX always grounded on the coax).
 
If somebody does measure a difference its down to mangled data on the USB interface (or perhaps poor measuring equipment - Mojo is way better than most test equipment). Mojo can't convert 16 bit data back to 24 bit....
 
Rob 

  .... Mojo has the same USB as Dave - but with Dave it is galvanically isolated, so RF noise and signal correlated currents from the source can't upset Dave at all. I can't do this with Mojo as it draws too much power from the device connected to the USB - but the upside is that mobile sources create much less noise as they are power efficient and there is no ground loops either due to battery operation.
 
So Mojo like Dave has solved the jitter problem via USB, as the timing for the data comes from Mojo not the source.
 
Rob

 
Also relevant:
 
 
 
 
Optical will support DSD with DoP on the optical - but only DSD64. Also, if you use a plastic fibre, only use very short lengths - for longer lengths you need a quality glass fibre. Running optical at 192 kHz is close to the edge for some optical transmitters and cables.

Rob

On a related note, for a given bitrate, is there any reason to expect optical cables of different build qualities to differ in SQ? Assuming they're capable of supporting the bitrate without noticeable artifacts, such as, intermittent "pops". Also, how big a role does the DAC implementation play?

Thank you for your opinion.

My listening test revealed no SQ change at all - so long as the data is arriving is still bit perfect. But with optical when it fails, it is fairly easy to spot bit failures. Of course, YMMV, and I guess if it's about to fail, you would hear an improvement with the odd bit error improvement. But I have not been able to hear a difference in my setup using plastic or glass.
 
Why would that be? Optical actually does not have bad jitter performance; but what it does do is have uneven rise and fall times. But my digital SPDIF receiver actually measures uneven rise and fall times, then uses that measurement to compensate to extract the data correctly. And as regards jitter - the DPLL completely removes jitter from the incoming stream - I can add 2uS worth of jitter, and see absolutely nothing coming out from Mojo with measurements - and Mojo's FFT noise floor is at -170 dB. So optical typically has 2nS of jitter, so that is a thousand times lower than a level that is still not detectable, even when I can resolve -170 dB.... So there is no technical explanation why it would make a difference. So if you do hear a difference, it is either because it is not bit perfect and has data errors (almost impossible with 44.1 though), or your suffering from a placebo (it looks nicer/costs more/must be better). 
 
Getting to your last point - the DAC has a big impact on this; most DAC's are very sensitive to jitter as they use analogue PLL techniques and they can't eliminate the jitter problems. So optical cables may have a SQ difference with other DAC's.
 
Rob

 
 

Is there a reason that different transports sound clearly different though, and more so on the mojo than on the Hugo? Not comparing between different inputs. The AK380 sounds clearly better to my ears than the AK100 does as as a transport. Among coaxial players the soundaware Esther m1pro sounds much better than most of the competition, and when the digital coaxial mode is activated and the amp and dac section are switched off it sounds even better.

Any possible thoughts on the reason for this?
smily_headphones1.gif

For electrical inputs - transports can make an audible difference for couple of reasons. RF noise from the source injected into the DAC ground plane will cause increased noise floor modulation; and the ear/brain is sensitive to minute levels of noise flloor modulation, so this is important - it will make it sound brighter with more noise floor modulation, and warmer and smoother with less. Additionally, depth and detail resolution is can be degraded by very tiny signal related but distorted currents; and this will subtly change small signal fundamental linearity (this is where small signals amplitude varies with signal level) and the ear/brain is incredibly sensitive to this; the smallest possible change in small signal resolution or accuracy will degrade the perception of depth. So very tiny distorted signal related currents will damage depth perception.
 
This is why optical is good; it does not suffer from any of these problems, as it is perfectly galvanically isolated.
 
And it explains why ASIO sounds better than WASAPI as less processor activity so less noise and hence better sound - so anything that you do with an electrical connection that reduces RF noise such as less processor activity or less power consumption may have a small benefit.
 
I have not noticed that Mojo is more sensitive than Hugo; if anything I would guess at maybe the other way around!
 
Rob

 
 
  Converting the original file into DSD or up-sampling is a very bad idea. The rule of thumb is to always maintain the original data as Mojo's processing power is way more complex and capable than any PC or mobile device.
 
DSD as a format has major problems with it; in particular it has two major and serious flaws:
 
1. Timing. The noise shapers used with DSD have severe timing errors. You can see this easily using Verilog simulations. If you use a step change transient (op is zero, then goes high) with a large signal, then do the same with a small signal, then you get major differences in the analogue output - the large signal has no delay, the small signal has a much larger delay. This is simply due to the noise shaper requiring time for the internal integrators to respond to the error. This amplitude related timing error is of the order of micro seconds and is very audible. Whenever there is a timing inaccuracy, the brain has problems making sense of the sound, and perceives the timing error has a softness to the transient; in short timing errors screw up the ability to hear the starting and stopping of notes.
 
2. Small signal accuracy. Noise shapers have problems with very small signals in that the 64 times 1 bit output (DSD 64) does not have enough innate resolution to accurately resolve small signals. What happens when small signals are not properly reproduced? You get a big degradation in the ability to perceive depth information, and this makes the sound flat with no layering of instruments in space. Now there is no limit to how accurate the noise shaper needs to be; with the noise shaper that is with Mojo I have 1000 times more small signal resolution than conventional DAC's - and against DSD 64 its 10,000 times more resolving power. This is why some many users have reported that Mojo has so much better space and sounds more 3D with better layering - and its mostly down to the resolving power of the pulse array noise shaper. This problem of depth perception is unlimited in the sense that to perfectly reproduce depth you need no limit to the resolving power of the noise shaper. 
 
So if you take a PCM signal and convert it to DSD you hear two problems - a softness to the sound, as you can no longer perceive the starting and stopping of notes; and a very flat sound-stage with no layering as the small signals are not reproduced accurately enough, so the brain can't use the very small signals that are used to give depth perception.
 
The second issue in using the transport to up-sample (44.1 to 176.4 say) is that the up-samplers in a PC or mobile device are very crude, with very limited processing power and poor algorithms. This results in timing problems, and like with DSD you can't hear the starting and stopping of notes correctly. These timing problems also screw up the perception of timbre (how bright or dark instruments sound), the pitch reproduction of bass (starting transients of bass lets you follow the bass tune), and of course stereo imagery (left right placement is handled by the brain using timing differences from the ears). Now Mojo has a very advanced algorithm (WTA) that is designed to maximise timing reconstruction (the missing timing information from one sample to the next) and huge processing power to more accurately calculate what the original analogue values are from one sample to the next. Its got 500 times more processing power than normal, and this allows much more accurate reconstruction of the original analogue signal.
 
So the long and the short is don't let the source mess with the signal (except perhaps with a good EQ program) and let Mojo deal with the original data, as Mojo is way more capable.
 
Rob

 
Also relevant:
 
Quote:
  It is always better to give Mojo bit perfect files and let Mojo do the work, as the processing within Mojo is much more complex and sophisticated than a mobile or PC.
 
So when you have an app that has a volume control, and no bit perfect setting, then set it to full volume on the app on the assumption that this will keep the data closer to the original file.
 
The volume control function on Mojo is much more sophisticated than the PC as I employ noise shaping and I do the function at a very high internal sample rate. Hopefully using the volume set to max on the app will mean the volume coefficient is 1.0000000... so it will return the original data.
 
Rob 

 
Quote:
  You can always do a listening test. If set to max against 50% say, and it hardens up (becomes brighter) with loud recordings then its clipping. If on the other hand the perception of sound-stage depth is reduced, then the volume control is degrading the sound.
 
If you do that test and can hear no difference then don't worry, its a good app volume control.
 
Rob
 
PS for fun I just did a very quick test using Dave. I listen to radio 3 using the BBC iPlayer. I normally have it set to max. I reduced the iPlayer volume control to half, boosted Dave volume control by +6dB - and yes I felt sound-stage depth was worse with lower iPlayer volume. 
 
 
 
 
 
So far I like using the Chord Mojo the most when listening to well-mastered vocalist stuff.
 
I think I get what reviewers and users mean when they remark that the soundstage seems narrower yet deeper when songs are played via the Mojo. I get a similar impression in the sense that the vocals seem more "forward". I don't mean sibilance or harshness, I mean the sense that the voice is being projected.
 
So I am really enjoying various CDs by vocalists. Happy to take a bit more time to now convert some of my fave vocalist albums to FLAC.
 
I also find the increase in detail and timing, especially when it comes to transients and drums, cymbals, hi-hats etc very impressive. But it's no something I want to emphasise too much because when listening to rock bands I wanna enjoy each song rather than concentrate on hearing the transients.
 
But really the Mojo is impressive for its price. Lots of times I get distracted cos I am like "wow! I never heard that detail before?!" and its a bit scary if I am walking on the street!

 
Perceived width is actually an aberration - so when image focus improves, the sensation of width diminishes. Its akin to looking at an image out of focus, then seeing it suddenly in focus - the size of the image gets smaller but you can see things much more accurately. Another way of looking at is perspective. If an instrument gets deeper into the sound stage, it naturally goes back and apparently decreases in width. So when you improve instrument placement focus in the sound stage, the perception of width will decrease.
 
There is an exception to this rule, in that you can encode sounds to sound wider than the loudspeakers, but this effect (replicating the phase delays and resonances of the earlobes and changing the left right phase) can increase width beyond the loudspeakers - it can also be used to encode height. But these effects are very rare and a bit hit and miss. Its these resonances and phase delays that allow binaural recordings on headphones to work. In this case, when image placement improves, then you get an increase in width and height - but as I said, these effects are very rarely found.
 
Rob

 
Also of interest:
 
  You can always do a listening test. If set to max against 50% say, and it hardens up (becomes brighter) with loud recordings then its clipping. If on the other hand the perception of sound-stage depth is reduced, then the volume control is degrading the sound.
 
If you do that test and can hear no difference then don't worry, its a good app volume control.
 
Rob
 
PS for fun I just did a very quick test using Dave. I listen to radio 3 using the BBC iPlayer. I normally have it set to max. I reduced the iPlayer volume control to half, boosted Dave volume control by +6dB - and yes I felt sound-stage depth was worse with lower iPlayer volume. 

 
 
  .... it is a 15T that is used on the Mojo.
 
That has 16,640 logic cells and 45 dsp cores. 44 cores are used in Mojo.
 
The overriding design decisions were about power consumption, so although more DSP cores are used than Hugo, that's to reduce power, as the DSP cores are run at a much lower clock speed. To give you another example of lower power, with Hugo when I needed a bigger multiplier I used one DSP core with FPGA fabric (logic cells) added to create the larger multiplier. With Mojo, to save power, I used multiple DSP cores and no fabric to create larger multipliers.
 
Only the WTA filter is different, the rest of the audio path has Hugo code.
 
Rob

 
Hugo and Dave don't use any kind of DAC chip, the analogue conversion is discrete using pulse array. The key benefit of pulse array - something I have not seen any other DAC technology achieve at all - is an analogue type distortion characteristic. By this I mean, as the signal gets smaller, the distortion gets smaller too. Indeed, I have posted before about Hugo's small signal performance - once you get to below -20 dBFS distortion disappears - no enharmonic, no harmonic distortion, and no noise floor modulation as the signal gets smaller. With Dave, it has even more remarkable performance - a noise floor that is measured at -180dB and is completely unchanged from 2.5v RMS output to no signal at all. And the benefit of an analogue character? Much smoother and more natural sound quality, with much better instrument separation and focus. Of course, some people like the sound of digital hardness - the aggression gets superficially confused with detail resolution - but it quickly tires with listening fatigue, and poor timbre variation, as all instruments sound hard, etched and up front. But if you like that sound, then fine, but its not for me.
 
On the digital filter front - original samples getting modified - actually the vast majority of FIR digital filters retain untouched the original samples, as they are known as half band filters. In this case, the coefficients are arranged so that one set is zero with one coefficient being 1, so the original sample is returned unchanged. The other set being used to create the new interpolated value. The key benefit of half band filters is that the computation is much easier, as nearly half the coefficients are zero, plus the filter can be folded so that the number of multiplications is a quarter of a non half band filter. When designing an audio DAC ASIC, the key part in terms of gate count is the multiplier, so reducing this gives a substantial improvement in die size, and hence cost. So traditional digital filters use a cascade of half band filters, each half band filter doubles up the oversampling - so a cascade of 3 half band filters will give you an 8 times over-sampled signal, with one sample being the unmodified original data. You can tell if the filter is like this as at FS/2 (22.05 kHz for CD) the attenuation is -6dB. The filters that are not like this are so called apodising filters, and my filter the WTA filter.
 
Going back eighteen years ago to the late 90's I was developing my own FIR filter using FPGA's. Initially, I was interested in increasing the FIR filter tap length as I knew from the mathematics of sampling theory that timing errors were reduced with increasing tap length. So the first test was to use half band Kaiser filters - going from 256 taps to 2048 taps gave an enormous sound quality improvement, so I had confirmed that tap length was indeed important subjectively. But at this point I was stuck; I knew that an infinite tap length filter with a sinc impulse response would return the original un-sampled signal perfectly - but the sinc function using only 16 bit accurate coefficients needs 1M tap FIR filter - and that would never happen, certainly not with 90's technology. So was it possible to improve the timing accuracy without using impossible tap lengths? After a lot of thinking and research, I thought there was a way - but it meant using a non half band filter, which would mean that the original sampled data would be modified. This was a big intellectual stumbling block - how can changing the original data be a good thing? But the trouble with audio is that neat simplistic ideas or preconceptions get in the way. Reality is always different, and reality can only be evaluated by a careful AB listening test. So I went ahead on this idea, and listened to the first WTA filter algorithm - and indeed it made a massive improvement in SQ - a 256 tap WTA sounded much better than 2048 tap half band Kaiser, even though the data is being modified. Why is this? The job of a DAC is NOT to reproduce the data it is given, but to reproduce the analogue signal before it is sampled. The WTA filter reconstructs the timing of the original transients much more accurately than using half band filters or filters that preserve the original data and it is timing of transients that is the most important SQ aspect.
 
So the moral of the tale? Don't let a simplistic technical story get in the way of enjoying music!                
  
 Rob

 
26,000 taps is the closest to a definitive statement as I've read ... the same as I've seen specified for Hugo.

I'm sure it's called out earlier in the thread, just going from memory as I'm far too lazy to search for it!

Actually it's about twice as many as Hugo but run at half the speed giving approximately the same number crunching power in terms of DSP ..... We have mentioned this before, but didn't want to put much focus on it as this is a small only part of the over design of Rob's overall topology

it was always our intention to try to match the performance of Hugo To do this without using as much power as Hugo. Therefore Rob used more DSP cores but run differently to match the performance of Hugo but at far lower power demands. JF

 
 
Quote:
  The earlier prototypes had less current than Hugo - 0.2A RMS. But last Feb we were with Nelson (Malaysian distributor) and I showed him the prototype. He brought some headphones, and we compared it to Hugo - and it was bad, a lot of clipping from Mojo, none from Hugo. It so happened (luckily) Nelson gave us possibly the only 8 ohm headphones on the market Final Audio Pandora.
 
So I upgraded the current to match Hugo, its the same 0.5A RMS. To do this I used 6 small OP transistors in parallel as size would not permit use of the large devices I use in Hugo.
 
Mojo development had some weird fortuitous events - the first time we showed the prototype happened to use the only 8 ohm headphones that would show problems - and the day we were deciding on the design Xilinx emailed me with a new FPGA that would enable Hugo performance but with the needed power, and happened to be in production just in time for Mojo. 
 
Quote:
   
1. The battery is capable of delivering 3A of current, and has very low impedance.
2. Mojo amplifier has a very high power supply rejection ratio.
3. The output is pure class A at 5v RMS into 300 ohm.
 
I just had an email today from a very experienced dealer that asked me this question:  
"Chord Mojo should have single amplification which drive to both headphone output, but if I'm using any headphones (HD800 for example, very heavy to drive) to put on headphone output 1 and I connect another headphone to headphone output 2 (Beyer T1 for example), I hear no differences on sound quality. Normally if one amplification section used to drive 2 headphones output, then once we connect the second headphone will make overall sound quality degradation (as the case of Beyerdynamic A20 amp, Grace Design M903, etc). What is the logical explanation of this? As it seems Mojo has 2 separate amplification sections which drive independently for each headphone output."
 
Of course Mojo does not have two amplification stages. It can drive two headphones with ease because it has exceptional low output impedance, and it has exceptional current linearity. So loading it with more headphones has no effect, unlike other headphone amplifiers.
 
Indeed, when I initially started designing Hugo I was shocked how poor from a measurement point of view headphone amps were. Poor output impedance, huge levels of distortion, and poor current linearity seem to be typical. And these things matter, if your goal is transparency and musicality. 
 
Rob
 
  I have just measured a Mojo into a 16 ohm load using an APX555 test equipment. With 1% THD 1 kHz single channel,  Mojo delivered 3.30 v RMS - that's 680 mW. Using 50 Hz, it was 668 mW RMS.
 
Rob

  I have done a quick measurement; with 30 ohms it is 4.25v RMS so that is 600 mW. For 50 ohms, I would expect 4.6v RMS or 423 mW RMS. This is with Mojo in blue battery, and at 1% THD with a continuous sine wave, power is RMS.
 
Rob

Quote:
  Into 300 ohms, fully charged battery, its 94 mW or 5.3v RMS at the 1% THD point.
 
Rob
 
 
Quote:
 
  Strange that the Hugo produces hiss (for those sensitive to it) with sensitive IEMs and the Mojo does not as the output specs are identical. If it were a lower powered Hugo aimed soley at IEM users, the lack of hiss would be expected, but it should be able to drive more demanding phones just as well. It is looking like a Hugo killer at a much lower price point.
The Hugo has a greater range of inputs and outputs and crossfeed for an additional £1000...

 
It's not strange if they used some kind of attenuator.

 
No attenuator as it would upset transparency.
 
I just reduced the noise - it is 125 dB dynamic range now. That said, "just reduced the noise" was not easy, it's one reason Mojo took so long to develop.
 
Rob
 
 
Quote:
 
Just a thought. Why do other DAC/headphone amps have amp sections when Hugo/Mojo get by without one, and many including Chord say it is more transparent? Have Chord got the patent for ampless amps :)


Because they can't using chip based DAC's. Chip DAC's have two current outputs. So you need two I to V converters (amps) then a differential to single ended amp, then a headphone buffer to deliver the current. You also need a lot of analogue filtering wrapped around these amps. So why are normal DAC's so complex in the analogue domain? Two reasons:
 
1. Silicon DAC's are horribly noisy, as the substrate and grounds are bouncing around due to switching activity. So to counter this, it is done differentially, which means the ground noise is cancelled. It also hides the problems of the reference circuitry, which can't be made with low enough impedance on silicon. This translates to more distortion, and crucially noise floor modulation.
 
2. Delta sigma converters run at low rates - best is at 12 MHz - this means that there is a lot of noise that must be aggressively filtered out in the analogue section. This also applies with R2R DAC's too as these have even worse problems due to the very slow switching speed.
 
So to run with a single amp section you need the DAC to be single ended and to run the noise shapers at much higher rates to reduce your filtering requirements. Because the analogue section with Mojo is discrete, I can use extremely low impedance and low noise reference supplies - something that is impossible on silicon. This has the other benefit of eliminating noise floor modulation (actually there is a lot more to it than this as there are countless other sources of noise floor modulation in a DAC). To make the filtering easier, the pulse array noise shapers run at 104MHz - over an order of magnitude faster than normal. There are other benefits to running the noise shapers at 104MHz, principally the resolving power of the noise shaper. Now soundstage depth is determined by how accurately small signals are reproduced. The problem with noise shaping is that small signals get lost - any signal below the noise shaper noise floor is lost information. But by running the noise shaper at much faster rates you solve this problem too - indeed Mojo noise shapers exceed 200dB THD and noise digital performance - that's a thousand times more resolving power than high end DAC's.
 
If I get time today I hope to publish noise floor modulation measurements showing Mojo has zero measured noise floor modulation. This level of performance does not happen on any other non pulse array DAC's at any price, and its the primary reason why Mojo sounds so smooth and musical.
 
Rob
 
  I have been seeing some comments describing Hugo as excellent DAC with a good headphone amp. Both comments, in my view, are wrong and way off the mark - and seeing these comments are starting to bug me, so I would like to get it off my chest. So forgive me if I am overstepping the mark - commenting on honest posts about a product I have designed, but I thought it might be useful for Head-fi'rs to read my views.
 
First, I would like to talk about what as a designer I am trying to accomplish, as it has a bearing on one's opinion of Hugo's sound. Imagine going around CES and carefully listening to all the high end hi-fi on show, so you can carefully listen to all the major high end brands available today. Next, listen center stage row 10 to an orchestra. Now, in my opinion, high end Hi-fi sounds from very bad to absolutely awful compared to live acoustic music. The key difference in the sound is variability - live acoustic music has unbelievable variations in the perception of space, timbre, dynamics and rhythm. Additionally, each instrument sounds separate and as distinct entities. By comparison, high-end audio is severely compressed - depth of sound stage is limited to a few feet (listen to off stage effects in say Mahler first - in a concert the off stage effects sound a couple of hundred feet away but on a hi-fi it is an ambient sound a few feet away). Timbre is compressed - you don't get a really rich and smooth instrument playing at the same time as something bright. The biggest problem is the dominance effect - the loudest instrument is the one that drags your attention away - this constant see-saw of attention is the biggest reason for listening fatigue, a major problem with Hi-fi.
 
So I am approaching designing of Hi-fi from the POV of accepting that there are enormous differences between conventional Hi-Fi and real music, and that I want my equipment to be as transparent as possible. Now some peoples idea of transparency is to use distortion to artificially enhance the sound, and this is a real problem with listening tests - a superficially brighter sound, giving the impression of better detail resolution, is often distortion. So a real challenge is defining what true transparency is. My definition, is to latch onto the idea of variations - if a modification makes the sound more variable, then its more expressive, and hence more transparent, even if it sounds, in tonal balance, darker or smoother and superficially less impressive. Now, if you think that your Hi-Fi sounds better than live acoustic music - then fine, we will agree to disagree. You are looking for a sculpted sound, not a truly transparent one, and I would strongly advise never to buy equipment designed by myself, as I am striving for equipment with no added sound.
 
So how does this relate to Hugo? Hugo was on the tail end of a long series of incremental improvements in digital design. I have spent the last 7 years on R and D to fundamentally improve aspects of DAC performance - improvements in the jitter rejection, RF noise filtering, noise shaper topologies, WTA filter length, analogue design plus a lot of other things. Moreover, Hugo took advantage of a big step forward in the capabilities of FPGA's - I could do important things that I knew influenced the sound but that previously were not possible due to FPGA limitations. So Hugo was at the confluence of two events - a big step forward from 7 years work in understanding digital design plus a major step forward in FPGA capability. It is just an accident that it happened with a portable headphone product.
 
So Hugo was the first instance when all these improvements came together. When I finally heard the pre-production unit with all the improvements in place I could not believe the sound quality improvements that I first heard. It completely changed my expectations of what was possible from digital audio - I was hearing things that I have never heard from Hi-fi ever - in other words, the gap from Hi-fi to live acoustic music was suddenly very much closer. Most notable was rapid rhythms being reproduced with breathtaking clarity - before piano music sounded like a jumble of notes, now I could hear each key being played distinctly. The next major change was timbre variations - suddenly each instrument had their own distinct timbre qualities, and the loudest instrument dominance effect was gone. Also gone was listening fatigue - I can listen for 12 hours quite happily.
 
But by far the biggest change was not sound quality, but on the musicality. I found myself listening and enjoying much more music, in a way I have never experienced before with a new design (and anybody who knows something of my designing career knows that is a lot of designs). 
 
So my conclusion is this: Hugo does things that no other DAC at any price point does. Now I can say readers saying, well OK he would say that anyway, it's his baby. True - I can't argue with that POV. But let's examine the facts:
 
1. The interpolation filter is key to recreating the amplitude and timing of the original recording. We know the ear/brain can resolve 4uS of timing - that is 250 kHz sampling rate. To recreate the original timing and amplitude perfectly, you need infinite tap lengths FIR filters. That is a mathematical certainty. Hugo has the largest tap length by far of any other production DAC available at any price.
 
2. RF noise has a major influence in sound quality, and digital DAC's create a lot of noise. Hugo has the most efficient digital filtering of any other production DAC - it filters with a 3 stage filter at 2048 FS. The noise shapers run at 104 MHz, some 20 times faster than all other DAC's (excepting my previous designs). What does this mean? RF noise at 1 MHz is 1000 times lower than all other DAC's, so noise floor modulation effects are dramatically reduced, giving a much smoother and more natural sound quality.
 
3. The lack of DAC RF OP noise means that the analogue section can be made radically simpler as the analogue filter requirements are smaller. Now in analogue terms, making it simpler, with everything else being constant, gives more transparency. You really can hear every solder joint, every passive component, and every active stage. Now Hugo has a single active stage - a very high performance op-amp with a discrete op-stage as a hybrid with a single global feedback path. This arrangement means that you have a single active stage, two resistors and two capacitors in the direct signal path -  and that is it. Note: there is no headphone drive. Normal high performance DAC's have 3 op-amp stages, followed by a separate headphone amp. So to conclude - Hugo's analogue path is not a simple couple of op-amps chucked together, it is fundamentally simpler than all other headphone amp solutions.
 
This brings me on to my biggest annoyance - the claim that Hugo's amp is merely good. Firstly, no body can possibly know how good the headphone amp in Hugo is, because there is not a separate headphone stage as such - its integrated into the DAC function directly. You can't remove the sound of the headphone amp from the sound of the DAC, it's one and the same.
 
Struck by these reports, I decided to investigate, as I see reported problems as a way of improving things in the future. I want to find weakness, my desire is to improve. So I tried loading the OP whilst listening on line level (set to 3v RMS). With 300 ohm, you can hear absolutely no change in sound. Running with 33 ohm, you can hear a small degradation - its slightly brighter. This is consistent with THD going from 0.0004% to 0.0007%. Note these distortion figures are way smaller than desktop headphone amps. Also note that with real headphones at this level you would be at typically ear deafening 115dB SPL. Plugging in real headphones (at much lower levels) gives no change in sound quality too. This has been reported by other posters - adding multiple headphones to Hugo does not degrade sound at all.
 
So how do we reconcile reports that desktop headphone amps sound better? I don't believe they do, its a case of altering the sound to suit somebody's taste. Now as I said at the beginning of this post, that is not what I want to do - I want things to sound transparent, so that we can get closer to the sound of live acoustic music. Adding an extra headphone amp will only make things worse as extra components degrades transparency. Another possibility is that people are responding against Hugo's unusually (for a headphone amp) low output impedance of 0.075 ohms. Now, compared to headphone amps of 2 to 33 ohms impedance, this will make the sound much leaner with less bass. Additionally, the improvements in damping can be heard as a much tighter bass with a faster tempo. So if you find your headphone too lean, the problem is not Hugo's drive - your headphone is just been driven correctly.                 
 
Just to close to all Hugo owners - enjoy! I hope you get as much fun from your music as I have done with Hugo. 

 
 

 
 
Quote:
 
  I found the reverse.  I'm using Sennheiser HD-25 1 II: directly out of the Mojo the sound seems present and correct, but when used with a Ray Samuels SR-71a, the sound goes to a whole new level.  The sound becomes rock solid and more like listening to musicians playing instruments; without the Ray Samuels the sound seem to collapse in on itself and become more hi-fi (ie impressive noises but less music).  To my ears, the extra amplification is not adding tonal euphony but is instead making the most of the DAC.
 
I have a theory that it's to do with the power supply: when using headphones more current is drawn and in a varying manner, ie it varies with the music.  This varying of current affects (I think modulates) the power supply voltage which affects the DAC, amplification and ultimately the sound. By connecting directly to an amp, there is less current drawn and no variation.  This might also explain why companies such as Naim claim improvements to their amps' sound quality when external power supplies are added.  Just my 2CW.


I do not buy this all. You need to bear in mind several facts:
 
1. The battery is capable of delivering 3A of current, and has very low impedance.
2. Mojo amplifier has a very high power supply rejection ratio.
3. The output is pure class A at 5v RMS into 300 ohm.
4. Reducing the output load only starts to increase distortion with 33 ohms - at this level it is very much lower than other headphone amps. The HD25 is a very easy 70 ohms.
5. Mojo is designed to drive loudspeakers. You will be amazed hearing it fill the room with beautiful sound using efficient 8 ohm horn loudspeakers.
 
Some people like the sound of more distortion - 2nd harmonic fattens the sound making everything sound phat, soft and rounded. But its not natural, nor do I find it musical, as everything sounds phat. I want soft sounds to sound soft, and sharp sounds to sound sharp - not everything to have a soft sheen on things all the time.
 
I can give you another example. I just had an email today from a very experienced dealer that asked me this question:
 
"Chord Mojo should have single amplification which drive to both headphone output, but if I'm using any headphones (HD800 for example, very heavy to drive) to put on headphone output 1 and I connect another headphone to headphone output 2 (Beyer T1 for example), I hear no differences on sound quality. Normally if one amplification section used to drive 2 headphones output, then once we connect the second headphone will make overall sound quality degradation (as the case of Beyerdynamic A20 amp, Grace Design M903, etc). What is the logical explanation of this? As it seems Mojo has 2 separate amplification sections which drive independently for each headphone output."
 
Of course Mojo does not have two amplification stages. It can drive two headphones with ease because it has exceptional low output impedance, and it has exceptional current linearity. So loading it with more headphones has no effect, unlike other headphone amplifiers.
 
Indeed, when I initially started designing Hugo I was shocked how poor from a measurement point of view headphone amps were. Poor output impedance, huge levels of distortion, and poor current linearity seem to be typical. And these things matter, if your goal is transparency and musicality. 
 
Rob
 
  There has been some talk about Mojo's hiss when silent. We publish the noise output voltage and its 3uV - that's same as an iPhone, and a little bit better than an AK240. With the Shure SE846 (pretty much the most sensitive IEM you can get) the 3uV translates to a noise of 24 dB SPL - and would be the same as the AK240 and the iPhone - but - and this is a big point - Mojo will also deliver over 5V RMS with the noise at 3uV still.
 
24dB will be audible to some, and not to others, as you naturally hear hiss with IEMs stuck in your ears. My K10's are completely inaudible with Mojo powered or not powered, similarly the ultimate ears UERM. But these devices work out at 6 dB SPL as they are sensibly sensitive.
 
Rob

  Just to reiterate on the hiss issue - Mojo has only 3uV of residual noise (that's the level with no signal). I have not seen a DAC, DAP, or mobile phone that betters this number, and this will determine the hiss level you hear. With sensible sensitivity IEM's (Noble, Ultimate ears, Dita) I can hear absolutely no added hiss from Mojo - that is turning Mojo on or off has no change in hiss levels from normal background noise.
 
Rob
 
  1. How does PSU design influence the sound quality of Rob's DACs?
 
  1. Detailed Blog on Listening Tests
 
 

 
 
(NB: please also view the VIDEOS section!)
 
 
 
www.the-ear.net/how-to/rob-watts-chord-mojo-tech
 
www.youtube.com/watch?v=3e7SRXP3RHI
 
Quote:
x RELIC x said:
/img/forum/go_quote.gif
  Q&A with Rob Watts:
 
 
Q: Were you able to fit the same tap filter length as the Hugo (26, 384 taps) with the same WTA filter in the Mojo?
 
A: "Mojo shares an extremely similar code as Hugo - the only change is the WTA filter is redesigned to accommodate 768 kHz. The new filter is broadly equivalent apart from this."
(Comment): When I pushed the tap length question with Chord they replied that "it will be a good while in the future before they publish this information, if at all". "The implementation within Mojo is different, but it’s not inferior to anything that we’ve done".
 

 

Q: In the Mojo presentation draft it mentions “Hugo like sound quality and musicality”. What differences in audio presentation would you say the Mojo has compared to the Hugo?
 
A: "Bearing in mind it’s use I have optimized the noise performance in order to make it sound smoother."
 

 

Q: The design for the Mojo began in 2012. Is it safe to say the Mojo R&D led to the Hugo until the technology caught up for the Mojo’s design target? Or, were they completely separate design goals?
 
A: “The R&D of Hugo and Mojo ran in parallel - the very first prototype (2012) was more like Mojo, then work switched to Hugo. Then I worked on Mojo in the background, with development getting really busy starting in Nov 2014. We built over 50 prototypes, as I had a lot of issues to contend with - thermals, charging, and getting SQ to be identical when charging were major headaches."
 

 

Q: Does the Mojo deal with jitter with the same DPLL as the Hugo?
 
A: “Yes, the DPLL is identical."
 

 

Q: I see the Mojo has an even better THD spec than the Hugo.
 
A: “Lower noise means better measurements."
 

 

Q: Is the Mojo analogue section Class A like the Hugo?
 
A: “The actual OP stage is identical - same OP transistor silicon - but I used 6 small transistors in parallel rather than 3 large devices. It’s biased at the same Class A level."
 

 

Q: Does Mojo have cross feed?
 
A: “No cross feed.”

 
 
Also see: Munkonggadgets interview with John Franks
 

 
Related discussion (not Mojo-specific, but much of it does apply to Mojo) on page 56 of this Rob Watts interview
 
...and this John Franks interview may be of interest, too.
 
Also, this one
 
 
 
and this: Interesting historical background of Rob's DAC design approach (video interview with Rob Watts & John Franks) (well worth watching)
 

 
 
  1. ohm-image.net/data/audio/rmaa-chord-mojo-24-bit
 
  1. goldenears.net/board/5904087
 
  1. stereophile.com/content/chord-electronics-mojo-da-headphone-amplifier-measurements
 
  1. hi-fiworld.co.uk/index.php/cd-dvd-blu-ray/62-cd-reviews/776-chord-mojo-review
 
 
 
  1. Quote:
  I promised some time ago that I would show some measurements showing Mojo's performance. My reasoning for this was that Mojo does things that no other (non Chord) DAC does at any price; I was kind of annoyed that some people were comparing it to $100 DACs when the true competitors were $100K - and I kind of get that, its difficult to take Mojo seriously given its size and price. But if you could see the design complexity that goes inside Mojo then one could appreciate how much better it is; it really is vastly more complex than other DAC's, and this complexity is needed to recreate the original analogue signal accurately.
 
But I can show you that something special is going on from measurements. Take a look at this plot. This is a FFT of a 1kHz output at 2.5v RMS into a 300 ohm load (blue trace) and then with no signal (red trace):
 

 
Now what is very interesting is the noise floor at -175dB - it does not change at all with 2.5v or nothing which indicates a complete absence of measurable noise floor modulation. Noise floor modulation is extremely important subjectively - you perceive the slightest amount as a brightness or hardness to the sound. When it gets bad, you hear glare or grain in the treble. All DAC's (apart from Chord DAC's) suffer from measurable noise floor modulation - typically the noise floor would be -160 dB with no signal, and -140 dB at 2.5v RMS. Some Class D amps are awful with noise floor at -120 dB (one reason why Class D often sounds so bad).
 
To get this measurement is a massive challenge, as ADC's themselves have large amounts of noise floor modulation, way bigger than my DAC's. The only test instrument that has noise floor modulation that can actually measure Mojo's performance is the APX555. This uses a novel approach to solving the issue - 4 ADC's and an analogue notch filters. The outputs are combined in the digital domain, so this means one ADC is handling the fundamental sine wave, another ADC looks at the noise via the notch filter. So you will only be able to measure Mojo's true performance using the APX555. 
 
Many posters have commented on how smooth and musical sounding Mojo is - and its in part down to the absence of measurable noise floor modulation. Actually getting this performance is very complicated, as within the DAC there are a enormous number of mechanisms to create noise floor modulation. One reason why its taken me 20 years of DAC development to do it!
 
Rob
 

 
 
(NB: please also view the VIDEOS section!)
 
 
  1. forbes.com/sites/marksparrow/2016/07/11/hi-res-audio-can-be-in-the-palm-of-your-hand-with-chords-mojo-dac-for-smartphone-users/#6b4011055255
 
  1. dailymail.co.uk/home/event/article-3690447/Chord-Mojo-flashback-days-British-hi-fi-single-best-audio-upgrade-buy.html
 
  1. whathifi.com/chord/mojo/review  (also see: whathifi.com/news/chord-electronics-dominates-best-dacs-2015)
 
  1. cnet.com/uk/news/chord-mojo-maximum-sound-quality-from-a-tiny-digital-converterheadphone-amplifier
 
  1. stereophile.com/content/chord-electronics-mojo-da-headphone-amplifier#m76BmGJUB2rti7gI.97
 
  1. telegraph.co.uk/luxury/technology/92513/chords-exceptional-audio-mojo.html (Ken Kessler)
 
  1. alphr.com/audio/1003966/chord-mojo-review-make-your-smartphone-sound-amazing
 
  1. metal-fi.com/chord-electronics-mojo/ (this contains quite an interesting discussion)
 
  1. the-ear.net/review-hardware/chord-electronics-mojo-portable-dacheadphone-amp
 
  1. audiovideo.fi/testi/chord-mojo-da-muunnin-kuulokevahvistin-testissa (in Finnish)
 
  1. stereo.net.au/reviews/review-chord-electronics-mojo-headphone-amplifier-dac
 
  1. artsexcellence.com/downloads/reviews/chord.mojo.artsexcellence.english.pd
 
  1. hifiplus.com/articles/chord-electronics-mojo-portable-dacheadphone-amp
 
  1. headphone.guru/the-chord-mojo-the-amazing-599-00-portable-wonder
 
  1. avforums.com/review/chord-mojo-dac-headphone-amp-review.12008
 
  1. digitaltrends.com/home-theater/chord-mojo-dac-amp-hands-on
 
  1. moon-audio.com/chord-mojo-dac-headphone-amp.html
 
  1. headfonics.com/2016/04/the-mojo-by-chord-electronics
 
  1. hi-fiworld.co.uk/index.php/cd-dvd-blu-ray/62/776.html
 
  1. howtospendit.ft.com/audiovisual/109351-chord-mojo
 
  1. blog.son-video.com/en/2016/09/review-chord-mojo
 
  1. headfonia.com/review-chord-mojo-the-chosen-one
 
  1. headfonia.com/review-chord-mojo-hot-or-not
 
  1. headphonescanada.ca/blog/chord-mojo-review
 
  1. head-fi.org/products/chord-mojo/reviews/14867
 
  1. head-fi.org/t/784602/10995#post_12328398
 
  1. head-fi.org/t/784602/5850#post_12110107
 
  1. head-fi.org/t/784655/chord-mojo-review
 
  1. custom-cable.co.uk/blog/chord-mojo-review
 
 
 
  1. "Now I finally understand the name. Chord’s got Mojo! Every audiophile needs his fix, and with Mojo he can get it anywhere."
 
  1. "A defining moment in audio reproduction. Very real holographic, 3d sound. The capabilities of this are literally awesome."
 
  1. "Chord Mojo Review - The Game Really Has Changed!" (NB: this review includes a useful short Q&A with Rob Watts)
 
  1. "A class leading sound quality DAC/AMP in a tiny footprint that work well with wide ranges of headphones."
 
  1. "The Chord Mojo is easily one of the most, if not the most outstanding product of 2015."
 
  1. "Mojo is very good at what it is intended to be used as – a portable DAC/amp"
 
  1. "Desktop Capable and Wholly Portable, A swiss Army knife in all but name!"
 
  1. "Mojo on the Go: A Review of Mojo and It's use from a Portable Perspective"
 
  1. "Great Value for Money. Possible Consideration for an All in One Solution."
 
  1. "Very musical. Great with FIIO digital output and USB output from a PC"
 
  1. "Chord Mojo : A small, affordable and highly musical portable device"
 
  1. "Absolutely the best portable amp/DAC combo on the market"
 
  1. "Chord Mojo: Top-notch sound in a small, portable device!"
 
  1. "Fulfillment of Foolish and Overwrought Expectations"
 
  1. "Excellent Mojo, A Unicorn at this price/performance"
 
  1. "The Chord Mojo: A Budget-Minded Rookie's Take"
 
  1. "A very accurate DAC/Headphone Amplifier"
 
  1. "CHORD Mojo - DAC/Headphone Amplifier"
 
  1. "A Lamborghini for the price of a Porsche"
 
  1. "Mojo Brings the Best Out of My IEMs"
 
  1. "MOJO: a little gem, highly musical"
 
  1. "Amazing sounding all in one unit!"
 
  1. "The Mojo. Get Inside Your Music"
 
  1. "GREAT HIGH END PRODUCT!!"
 
  1. "A great little portable device"
 
  1. "Audiophile Basshead grade"
 
  1. "The Magical Black Box"
 
  1. "Big Bang, Little Box"
 
  1. "An Instant Classic"
 
  1. "I found my MoJo"
 
  1. "Excellent dac"
 
  1. "Chord Mojo"
 
 

 
"We use it all the time..even for testing internally. It sounds awesome."
 
www.head-fi.org/t/784602/20595#post_12753517
 
www.head-fi.org/t/784602/26865#post_13061577
 
www.head-fi.org/t/784602/15165#post_12472084
 
www.head-fi.org/t/784602/15390#post_12480734
 
www.head-fi.org/t/784602/11055#post_12330962
 
www.head-fi.org/t/784602/16260#post_12516947
 
www.head-fi.org/t/784602/19365#post_12691852
 
www.head-fi.org/t/784602/18495#post_12634409
 
www.head-fi.org/t/784618/chord-mojo-impressions-thread
 
www.head-fi.org/t/784602/12105#post_12381029 (also discusses an innovative stacking approach)
 
www.head-fi.org/t/784602/13620#post_12430618
 
www.head-fi.org/t/784602/1140#post_12007799
 
www.head-fi.org/t/784602/15630#post_12492483
 
www.head-fi.org/t/784602/11775#post_12363801
 
www.head-fi.org/t/784602/1185#post_12008368
 
www.head-fi.org/t/784602/6825#post_12161630
 
www.head-fi.org/t/784602/1440#post_12011978
 
www.head-fi.org/t/784602/1470#post_12012541
 
www.head-fi.org/t/784602/1650#post_12015292
 
www.head-fi.org/t/784602/1755#post_12016908
 
blog.moon-audio.com/chord-mojo-review/
 
www.digitalaudioreview.net/2015/10/chord-electronics-mojo-portable-audios-new-talisman/
 
gavinsgadgets.com/2015/10/23/the-chord-mojo-the-game-changer-has-arrived-first-impressions/
 
www.hifiplus.com/articles/first-look-chord-mojo-portable-dacheadphone-amp/
 
www.digitalaudioreview.net/2016/02/chord-electronics-mojo-lost-found/
 
www.head-fi.org/t/784602/1950#post_12019292
 
www.head-fi.org/t/784602/13110#post_12417346
 
www.head-fi.org/t/784602/2025#post_12020114
 
www.head-fi.org/t/784602/2100#post_12021153
 
www.head-fi.org/t/784602/2535#post_12028236
 
www.head-fi.org/t/784602/3165#post_12038719
 
www.head-fi.org/t/784602/3360#post_12041678
 
www.head-fi.org/t/784602/3675#post_12048000
 
www.head-fi.org/t/784602/3855#post_12050914
 
www.head-fi.org/t/784602/4620#post_12065951
 
www.head-fi.org/t/784602/4665#post_12067288
 
www.head-fi.org/t/784602/4740#post_12068725
 
www.head-fi.org/t/784602/4800#post_12070256
 
www.head-fi.org/t/784602/4815#post_12070908
 
www.head-fi.org/t/784602/4815#post_12070955
 
www.head-fi.org/t/784602/5010#post_12077101
 
www.head-fi.org/t/784602/5475#post_12094314
 
www.head-fi.org/t/784602/5565#post_12096634
 
www.head-fi.org/t/784602/5805#post_12106201
 
www.head-fi.org/t/784602/5775#post_12103485
 
www.head-fi.org/t/784602/5820#post_12106718
 
www.head-fi.org/t/784602/5835#post_12107978
 
www.head-fi.org/t/784602/5880#post_12111500
 
www.head-fi.org/t/784602/5880#post_12112325
 
www.head-fi.org/t/784602/5970#post_12116364
 
www.head-fi.org/t/739712/1964-ears-adel-iems/2835#post_12137561
 
www.head-fi.org/t/784602/6210#post_12131983
 
www.head-fi.org/t/784602/6450#post_12145215
 
www.head-fi.org/t/784602/6615#post_12151064
 
www.head-fi.org/t/784602/6705#post_12154267
 
www.head-fi.org/t/784602/6720#post_12155146
 
www.head-fi.org/t/784602/6735#post_12155440
 
www.head-fi.org/t/784602/15960#post_12505605
 
www.head-fi.org/t/784602/7035#post_12172329
 
www.head-fi.org/t/784602/7050#post_12172799
 
www.head-fi.org/t/784602/7050#post_12173127
 
www.head-fi.org/t/784602/7095#post_12173591
 
www.head-fi.org/t/784602/7455#post_12187762
 
www.head-fi.org/t/784602/7890#post_12203042
 
www.head-fi.org/t/784602/7905#post_12203566
 
www.head-fi.org/t/784602/9570#post_12268684
 
www.head-fi.org/t/784602/9660#post_12272381
 
www.head-fi.org/t/784602/9690#post_12273665
 
www.head-fi.org/t/784602/9690#post_12273890
 
www.head-fi.org/t/784602/9780#post_12276415
 
www.head-fi.org/t/784602/10095#post_12291355
 
www.head-fi.org/t/784602/10575#post_12309334
 
www.head-fi.org/t/784602/10650#post_12313993
 
www.head-fi.org/t/784602/10650#post_12314170
 
www.head-fi.org/t/784602/11115#post_12333070
 
www.head-fi.org/t/784602/11775#post_12363723
 
www.head-fi.org/t/784602/11820#post_12366263
 
www.head-fi.org/t/784602/12675#post_12404483
 
www.head-fi.org/t/784602/12705#post_12405102
 
www.head-fi.org/t/784602/12795#post_12407006
 
www.head-fi.org/t/784602/12795#post_12407032
 
www.head-fi.org/t/784602/14430#post_12452128
 
 

 
 

Other relevant posts and threads:

 
 
Just a thought. Why do other DAC/headphone amps have amp sections when Hugo/Mojo get by without one, and many including Chord say it is more transparent? Have Chord got the patent for ampless amps :)


Because they can't using chip based DAC's. Chip DAC's have two current outputs. So you need two I to V converters (amps) then a differential to single ended amp, then a headphone buffer to deliver the current. You also need a lot of analogue filtering wrapped around these amps. So why are normal DAC's so complex in the analogue domain? Two reasons:
 
1. Silicon DAC's are horribly noisy, as the substrate and grounds are bouncing around due to switching activity. So to counter this, it is done differentially, which means the ground noise is cancelled. It also hides the problems of the reference circuitry, which can't be made with low enough impedance on silicon. This translates to more distortion, and crucially noise floor modulation.
 
2. Delta sigma converters run at low rates - best is at 12 MHz - this means that there is a lot of noise that must be aggressively filtered out in the analogue section. This also applies with R2R DAC's too as these have even worse problems due to the very slow switching speed.
 
So to run with a single amp section you need the DAC to be single ended and to run the noise shapers at much higher rates to reduce your filtering requirements. Because the analogue section with Mojo is discrete, I can use extremely low impedance and low noise reference supplies - something that is impossible on silicon. This has the other benefit of eliminating noise floor modulation (actually there is a lot more to it than this as there are countless other sources of noise floor modulation in a DAC). To make the filtering easier, the pulse array noise shapers run at 104MHz - over an order of magnitude faster than normal. There are other benefits to running the noise shapers at 104MHz, principally the resolving power of the noise shaper. Now soundstage depth is determined by how accurately small signals are reproduced. The problem with noise shaping is that small signals get lost - any signal below the noise shaper noise floor is lost information. But by running the noise shaper at much faster rates you solve this problem too - indeed Mojo noise shapers exceed 200dB THD and noise digital performance - that's a thousand times more resolving power than high end DAC's.
 
Mojo has zero measured noise floor modulation. This level of performance does not happen on any other non pulse array DAC's at any price, and its the primary reason why Mojo sounds so smooth and musical.
 
Rob

 
Quote:
  the OP stage is integrated with the OP filter. This means that Mojo analogue section is very simple, so giving Mojo's transparency, but the downside is a small variation in frequency response with load impedance.
 
Rob
 
Quote:
 
  .... the Mojo on line level mode - does this still run thru the Mojo's amp? from how i understand your earlier descriptions, buth the amping and DAC is done in the FPGA?
thus there is no way to truly use it as a dac without double amping?

Line level mode is just a volume preset for the volume control - nothing else changes.
 
Mojo has an FPGA (which is digital logic only) a discrete DAC (turning digital signals to analogue via flip-flops and resistors) and a single output amplifier - and that is it.
 
Conventional DAC headphone amps use differential outputs and have two I to V converters (current to voltage), a differential to single ended converter, and an output amplifier. Wrapped up with that is a analogue filter. So that's a lot of passive components and four amplifiers in the signal path. 
 
Because Mojo's FPGA has extensive digital filtering (at 2048 FS) and has a noise shaper that runs at a very high rate (104MHz) and uses a discrete DAC, I can keep the analogue section radically simpler, and this is one reason why Mojo is so transparent compared to all other DAC amps.
 
Rob

 
  @xtr4 i understand the FPGA designs makes the dac and amp essentally the same... what im really trying to get at is, can the FPGA's amp functions be bypassed so it is used simply as a DAC, and the two 3.5mm outs are true line outputs to prevent double amping
Paste

No, you need at least one amplifier to do the critical I to V conversion. Now it is possible to design a voltage only DAC (no amp at all), but they sound poor due to lots of problems - the largest being the huge amount of distortion you get doing it that way. Believe me, if I could make it simpler I would. The key that Mojo has is extremely low distortion and noise (0.00017% 3V 300 Ohms) but only one single amplifier in the signal path - and this amp combines headphone drive, filtering and I to V conversion in a single stage.
 
Rob

 


 
 
 
Of course the balanced output is going to be better than the Mojo, the Mojo doesn't have balanced output.

No that simply is not correct! A single ended design, done right with a large enough voltage swing will easily out perform a balanced output. Balanced designs are used by some designers to overcome inherent limitations within designs. Usually to overcome substrate noise on the chip that shouldn't be there or to increase the output voltage swing of their amplifiers. We don't suffer those limitation or problems so we don't need a dodgy fix for them. Our measurements clearly show this. Sorry to burst you bubble man.

Balance operation is a fix for problems we don't have. We have no substrate noise and we have plenty of output swing. Single ended done right is far better than a balanced design far less distortion.

 
Quote:
Originally Posted by Rob Watts /img/forum/go_quote.gif
Quote:
Originally Posted by agisthos /img/forum/go_quote.gif
Rob you should give a definitive 'why SE is better' explanation. Get it over with, because many (most) audiophiles have been biased towards balanced and are not going to understand where you are coming from.
 
One good argument I heard from the Densen founder (Thomas Sillesen) is that each half of the signwave runs through a series of components that will always have tolerances different from each other, so when combining the signal they will not ever match, causing an increase in distortion (of some kind I cannot remember).
 
Charles Hanson, of Ayre, who is a proponent of fully balanced equipment, has even stated that for pure sound quality SE will always sound better, but this is on the bench, where the power supply and analog signal stages can be kept physically apart. When putting them in a box he prefers balanced.

Well this is a complex subject, and sometimes a balanced connection does sound better than single ended (SE) - in a pre-power context - but it depends upon the environment, and the pre and power and the interconnect. But the downside of balanced is that you are doubling the number of analogue components in the direct signal path, and this degrades transparency. In my experience every passive component is audible, every metal to metal interface (including solder joints - I once had a lot of fun listening to solder) has an impact - in case of metal/metal interfaces it degrades detail resolution and the perception of depth. So going balanced will have a cost in transparency.
 
In DAC design, going balanced is essential with silicon design; there is simply too much substrate noise and other effects not too. But with discrete DAC's you do not need to worry about this, so going SE on a discrete DAC is possible, and is how all my DAC's are done. But differential operation hides certain problems (notably reference circuit) that has serious SQ effects; so going SE means those problems are exposed, which forces one to solve the issue fundamentally. In short, to make SE work you have to solve many more problems, but the result of solving those problems solves SQ issues than differential operation hides when you do measurements.
 
Rob 

Quote:
Originally Posted by Rob Watts /img/forum/go_quote.gif
Component count is very important for transparency. Doubling the number of parts in the direct signal path does degrade depth perception and detail resolution.
 
But there is another problem with balanced operation. Imagine a balanced differential in, differential out amplifier. The input stage is normally a differential pair (maybe cascoded) with a constant current source. Now the input stage is free to move up and down to accommodate the common mode voltage - but the input stage common mode impedance is non linear, and if the common mode voltage has a signal component (it always will have due to component tolerances) then this will create a signal dependent error current, thereby generating distortion. Unfortunately, the negative feedback loop of the amplifier can't correct for this distortion as it can't see the error on the summing nodes. So there will always be a limit to the performance. With SE operation, this problem does not occur, as the differential input stage is clamped to ground.
 
Now DAC designers are well aware of this - that's why all high performance DAC's use two single ended I to V converters from the current OP of the DAC's, then use a differential to SE converter to create the voltage OP. There are other reasons for doing this as well, as the DAC requires a very low impedance virtual ground for low distortion, and you can only get this using dual SE amps - another problem is RF and its much easier to decouple SE than differentially - this in turn creates a lot more noise floor modulation, making it sound less smooth.
 
But for me the most important is transparency. I had an amp that had two modes - differential or SE - listening in balanced mode flattened the sound stage depth dramatically,and it sounded harder, less smooth. That said, there are circumstances when balanced operation can be better than SE, for example when you are looking at connecting a pre-amp to a power amp, and what is best depends upon particular circumstances. In short, if SE operation is noisy, try balanced.
 
Rob 

 
 
 
 
@robwatts @mojo ideas

How about an impedance module that allowed us to adjust output impedance until we perfectly matched mojo to our ciems/headphones?

The technically perfect impedance is zero, and that's why I worked so hard to get it as low as 0.075 ohms with Mojo.
 
The reasons going for as close as zero are:
 
1. Frequency response. The impedance of the headphone varies with frequency, and so by having a high output impedance will cause frequency response variations. Zero impedance eliminates this problem.
 
2. Distortion. The impedance of a headphone varies with level, and having a higher output impedance will increase the total distortion - given that Mojo distortion is so low, this is actually quite a significant an effect. Again, zero impedance eliminates this problem.
 
3. Damping factor - probably the most important reason. A drive unit is a resonant system - that is a mass on a spring - that is damped mechanically and electrically. Electrical damping is due to the headphone creating a current due to the motion of the driver in the magnetic field - and how well this is controlled depends on the electrical impedance the driver sees - in our case, the cable impedance and Mojo's impedance. Again, zero impedance gives the best damping, with an infinite damping factor.
 
I did some listening tests many years ago with loudspeakers and damping factor and found that it made a massive difference to the sound. Damping of 10 gave a very soft, big fat bass - but everything sounding one note in the bass - simply because the loudspeaker was doing its own thing at the resonant frequency. Going from 10 to 100 gave a tighter bass, with much better pitch reproduction - you could follow the bass line much more easily. Above 100 to 1000 it sounded tighter - no big change in pitch (being able to follow the bass tune) but the perceived tempo of the music became faster as transients are much better controlled. Going above 1000 gave a small improvement in how tight it sounded.
 
Rob

 
 
 
Please note that the following quote was posted in the 2qute DAC thread, and is referring to the Hugo, so please exercise some discretion in that the Hugo is not 100% identical to the Mojo, but the majority of this information does apply equally-well to Mojo:
 
 
  Dear Rob
 
What is a OP stage? I understand discrete stage is better than op-amp, could you explain why? As I understand the Hugo has no analog volume control, so the output from the DAC doesn't go through a preamp (like one of the competing products from Salisbury)
 
Also what is a pulse array dac? is it similar to Delta Sigma or the resistor ladder Dac?  Is the sound of the hugo due to the filter or due to filter/dac combination? Also if you were to use this filter with a conventional resistor ladder DAC would it work?
 
Thanks
Analog

Welcome to Head-Fi analogmusic, and I am pleased you are enjoying more musicality from your music with Hugo - which is what this is all about!
 
What is an OP stage?
OP is output, and it replaces rather poor OP stages within op-amps. When faced with designing the electronics of Hugo, I had no experience of designing headphone amps - looking into devices that supplied headphones, they were very poor. So I designed it as if it was a power amp (I've designed lots of those) and gave Hugo the ability to drive 8 ohm loudspeakers directly - which means lots of current - in Hugo's case I set it too 0.5A RMS. You will not get this current from op-amps or headphone drive chips, so I had to design a discrete amp. Now to get the best transparency there needs to be a single feedback path, so the discrete OP stage needs to be within the op-amp's global feedback path. Since the op-amps are very high gain bandwidth product devices (high speed), that meant designing a Class A OP stage with very low propagation delay, so that the circuit would remain stable. Now the op-stages in op-amps are pretty poor to awful, so when I got the first prototype I was very pleased at how good the OP stage sounded, and how much lower distortion was (particularly high order harmonics) - even when using the op-stage in DAC mode with easy loads. Indeed, I now use this arrangement all the time now, as it really improves the performance of the op-amp - that's why 2 Qute has it too. The OP stage is by far the weakest part of all op-amps and this is simply because one can use a decent Class A bias current, and very substantial OP transistors, so thermal stability is ensured. And yes, Hugo does not have an analogue volume control, so this means the analogue section is very simple (just 2 resistors and capacitors in the direct signal path). Simple analogue gives much more transparency.
 
What is a Pulse Array DAC?
This is not an easy answer, as its complex and of course proprietary. But firstly the history. I first started designing DAC's in 1989, when the first delta-sigma bitstream devices from Phillips came out - these were DSD 256 DAC's (or PDM dac's). Now they were quite musical, but had technical and SQ problems - but they had very good low signal performance, and analogue distortion characteristic (small distortion for small signals unlike R2R DAC's which have more distortion for small signals due to glitch energy and resistor matching problems - issues that are impossible to solve). The biggest problem was limiting of resolution - unlike PCM, where ultra small signals are buried in the dither and so perfectly preserved, with delta-sigma the noise floor is a cliff edge for low level signals - any small signal below the resolving power of the noise shaper is lost forever. To overcome this, I used 8 PDM noise shapers with different dither, and summed the output in the analogue current to voltage converter (I to V). This gave much better performance, but I knew that much more was possible. So I started creating my own noise shapers and DAC technology using FPGA that were just becoming available (1994 now). What I needed was much higher resolution so the noise shaper OP is 5 bits not 1 bit, and I ran the noise shapers at a much higher rate - 2048 times not 256 times. Running at a faster rate means that you have more permutations of OP, which translates to much better performance. Run a 5th order noise shaper at ten times the speed, you can get in the digital domain, up to 100 dB lower distortion and noise - that's a 100 dB improvement in small signal resolution, so running at much higher rates gives massive improvements in SQ and measurements. Twenty years on, and I am still the only silicon/FPGA DAC designer running as high as this rate - delta-sigma DAC's are still stuck at 256 times or below.
 
But changing from single bit to multi-bit noise shaping may throw the baby out with the bathwater. The primary benefit of single bit is that it can (if you are very very careful) have zero small signal distortion, as there are no resistors to balance, as there is only one. With 16e Pulse Array, there are 16 PWM elements, and each element has on the long term exactly the same data, but instantaneously slightly different data. The benefit of the Pulse Array scheme is that when the elements are slightly different in value, it creates a fixed signal independent noise, and absolutely no distortion, but has innately higher resolution of 5 bits. That's why Hugo has (uniquely compared to other non Pulse Array DAC's) no measurable distortion, or any other artifact, for signals below -30 dBFS (see plots in previous posts). Additionally, because of the way the array is composed, master clock jitter has no significant affect - random jitter gives a tiny insignificant fixed noise. Its why I don't go endlessly on about femto clocks as the DAC is innately jitter insensitive. There are many more problems with noise shaping, as it is a very complex subject, but this will give you a flavour of the issues involved.  
 
Is the sound of the hugo due to the filter or due to filter/dac combination?
The sound of Hugo is down to lots of things, but of course the primary problem that Hugo addresses is the time domain one. That's where we are converting the sampled data into the original un-sampled continuous analogue waveform - the original signal at the ADC sampling point. Now we are trying to re-create the original un-sampled waveform - re-creating all the missing bits of data from one sample to the next one. Now the theory is very straightforward - if you use an infinite tap length FIR filter with a sinc impulse response you will absolutely and perfectly reconstruct the bandwidth limited signal - if its perfectly bandwidth limited to below 22.05 kHz it will not matter if you sample at 22 uS or 22 femtoS it will make no difference to the output - if you use an infinite tap length FIR filter. Now of course, we can't have infinite tap lengths filters, we have to make do with something very limited.
 
The question is, what level of time domain accuracy do we need where improving it makes no difference to the sound quality? That's where lots of careful listening tests comes in, as nobody knows. And its where I have been spending a lot of time over the last 18 months working on project xxxx - and I have learnt a lot (and I still have more things to discover, I am sure that I have not gotten to the bottom of the time domain accuracy barrel). What is clear to me, is that the ear/brain is amazingly sensitive to tiny time domain errors - there does not seem to be a level which one can say is insignificant. This is one of the really weird and interesting things about correlating what one hears with real signal errors - the other really odd issue being the perception of sound-stage depth - this can be upset by seemingly impossibly small errors.
 
This is where I find the "DAC bit perfect" concept  - like a cheap politicians sound byte - ridiculous. The job of a DAC is to reproduce the continuous waveform at the ADC sampler - NOT to bit perfectly reproduce the sampled data with all the sampling time domain errors perfectly intact.  
  
If you were to use this filter with a conventional resistor ladder DAC would it work? 
The answer to this is yes, but not as well as Pulse Array - the 16e DAC can reproduce 50 MHz sine wave albeit with 3% THD and noise! The problem with R2R is that the OP can't switch fast enough, as there are a lot of switches involved in the R2R ladder, so in practice you can't run them above 16 FS - but I can run mine at 2048 FS so the digital domain is much closer to the original un-sampled analogue waveform. There are lots of other problems with R2R - noise floor modulation, code dependent glitch energy, high distortion at small signal levels, and moderate distortion at large signal levels.
 
 
I hope I have not confused things too much - but we are dealing with a very complex subject, and something which, after more than 30 years of intense work, I am still learning new things. Things are very complex when you dive into it, and the ear/brain is a remarkably sophisticated device - the illusion of listening to real sounds is a truly amazing brain construct, and its something we know very little about. But at the end of the day, the engineering that goes into Hugo does not matter, its the musicality that counts, so keep on enjoying music! 
 
Rob

 
Quote:
  The earlier prototypes had less current than Hugo - 0.2A RMS. But last Feb we were with Nelson (Malaysian distributor) and I showed him the prototype. He brought some headphones, and we compared it to Hugo - and it was bad, a lot of clipping from Mojo, none from Hugo. It so happened (luckily) Nelson gave us possibly the only 8 ohm headphones on the market Final Audio Pandora.
 
So I upgraded the current to match Hugo, its the same 0.5A RMS. To do this I used 6 small OP transistors in parallel as size would not permit use of the large devices I use in Hugo.
 
Mojo development had some weird fortuitous events - the first time we showed the prototype happened to use the only 8 ohm headphones that would show problems - and the day we were deciding on the design Xilinx emailed me with a new FPGA that would enable Hugo performance but with the needed power, and happened to be in production just in time for Mojo. 
 
Quote:
  I have just measured a Mojo into a 16 ohm load using an APX555 test equipment. With 1% THD 1 kHz single channel,  Mojo delivered 3.30 v RMS - that's 680 mW. Using 50 Hz, it was 668 mW RMS.
 
Rob

  Into 300 ohms, fully charged battery, its 94 mW or 5.3v RMS at the 1% THD point.
 
Rob

  1. The battery is capable of delivering 3A of current, and has very low impedance.
2. Mojo amplifier has a very high power supply rejection ratio.
3. The output is pure class A at 5v RMS into 300 ohm.
4. Reducing the output load only starts to increase distortion with 33 ohms - at this level it is very much lower than other headphone amps. The HD25 is a very easy 70 ohms.
5. Mojo is designed to drive loudspeakers. You will be amazed hearing it fill the room with beautiful sound using efficient 8 ohm horn loudspeakers.
 
Some people like the sound of more distortion - 2nd harmonic fattens the sound making everything sound phat, soft and rounded. But its not natural, nor do I find it musical, as everything sounds phat. I want soft sounds to sound soft, and sharp sounds to sound sharp - not everything to have a soft sheen on things all the time.
 
I can give you another example. I just had an email today from a very experienced dealer that asked me this question:
 
"Chord Mojo should have single amplification which drive to both headphone output, but if I'm using any headphones (HD800 for example, very heavy to drive) to put on headphone output 1 and I connect another headphone to headphone output 2 (Beyer T1 for example), I hear no differences on sound quality. Normally if one amplification section used to drive 2 headphones output, then once we connect the second headphone will make overall sound quality degradation (as the case of Beyerdynamic A20 amp, Grace Design M903, etc). What is the logical explanation of this? As it seems Mojo has 2 separate amplification sections which drive independently for each headphone output."
 
Of course Mojo does not have two amplification stages. It can drive two headphones with ease because it has exceptional low output impedance, and it has exceptional current linearity. So loading it with more headphones has no effect, unlike other headphone amplifiers.
 
Indeed, when I initially started designing Hugo I was shocked how poor from a measurement point of view headphone amps were. Poor output impedance, huge levels of distortion, and poor current linearity seem to be typical. And these things matter, if your goal is transparency and musicality. 
 
Rob

Charging state of the battery makes little difference to the output level

 
 

 
Regarding MFI certification (Mojo)
 

IMPORTANT!: Please be careful with the iOS public betas - they can cause crackling sounds when using Mojo with the CCK
   
(NB: please also view the VIDEOS section!)
 
IMPORTANT : It is YOUR responsibility to make sure that the cable you are buying correctly fits the connectors on your equipment. None of these links are official endorsements.
 

 
Mojo is not Apple MFI-certified (it doesn't have an Apple CCK/MFI chip integrated inside), so connecting Mojo to an iDevice requires a cable with an included MFI chip. This is generally a CCK Lightning to USB Camera Adapter (which must be linked to the male-USB-to-male-microUSB cable that's included with Mojo):

 
I believe a requirement of using the camera kit Apple chip inside your product is complete design disclosure to Apple engineering hardware software the lot. We might be a bit mad but we're not totally crazy. If we had not much technology to hide. Say if we were just using a industry Dac chip we wouldn't have a problem in doing this, but for us today it's a very different story. This is why we will soon offer a plug in module that swallows the official Apple Camera adaptor leaving just the Lightning tail to plug into the I phone.

Apple CCK is a must unless you have a specialty cable with the Apple MFI chip inside.

Your connection should be this:
 
   
.... There is no Apple Co-processor in the Hugo or Mojo so an Apple CCK connector is needed. We (Moon Audio) are MFI certified to build Apple Lightning Cables but Apple will not let anyone build an all in one Lightning CCK cable. trust me we have tried and asked for all kinds of variances on this. This is how Apple makes money on Licensing Co-processor to Dac manufacturers. If we introduce a cable that solves this that reduces the bank role of Apple. Chord will have several accessories down the road the plug into the end of the Mojo, one of which will swallow the end of a CCK. See my Mojo review here of pics: http://blog.moon-audio.com/chord-mojo-review/

 
 
Moon-Audio's video overview of connecting Mojo with Apple's CCK cable
 
 
If you dislike using the Apple CCK connector, there are some 3rd-party cables which circumvent this, and thereby allow you to use a single cable. However, they are not Apple-certified, so there is a small risk that Apple may find a way to stop them working, in a future iOS update. At this point in time, though (iOS 9.3), many people are successfully using them. Ultimately, please do your own research before buying/trying any of these CCK-circumvention cables!!
 
Lavricable (circumvents Apple CCK)
 
CAUTION!  This cable does not appear to function well with iOS 10, so probably best to choose an alternative until (IF) the issue is resolved

 
lavricables@gmail.com
www.ebay.com/itm/Pure-Solid-Silver-Lightning-to-Chord-Hugo-Mojo-interconnect-cable-Iphone-4-5-6-/172223242191
 
 
ZY Cable Lightining to MicroUSB cable (circumvents Apple CCK)
 
WARNING! - Although this (seemingly-generic) cable has worked for many people, reports have begun to come in, during Jan 2017, that it may now not be working
 
http://www.head-fi.org/t/784602/chord-mojo-dac-amp-faq-in-3rd-post/28275#post_13142121
http://www.head-fi.org/t/784602/chord-mojo-dac-amp-faq-in-3rd-post/28530#post_13150044
 
This looks to be identical to the Hi-FiSpot cable, listed beneath this one
 
CAUTION!  The thin braided wire used in this cable may not be the most resilient to wear&tear
 
Quote:
Just to close the loop, I received ZY Cables' Lightning to MicroUSB cable yesterday (along w/ AQ Nighthawks! :)). Purchased here. So far, no issues on iPhone 7+ 128GB running iOS 10.1.1. Sample rate reflected correctly on Mojo when playing high-res from Onkyo HF, NePlayer and HibyMusic. Seems to work well (have been running in Airplane mode so don't know about RF rejection). Will see what happens with the next iOS update... Thanks all.
 

 

 

Available from:
www.ebay.com/itm/112101143986
 
Hi-Fi Spot Lightning Cable
 
WARNING! - Although this (seemingly-generic) cable has worked for many people, reports have begun to come in, during Jan 2017, that it may now not be working
 
http://www.head-fi.org/t/784602/chord-mojo-dac-amp-faq-in-3rd-post/28275#post_13142121
http://www.head-fi.org/t/784602/chord-mojo-dac-amp-faq-in-3rd-post/28530#post_13150044
 
This looks to be identical to the ZY cable, listed above
 
CAUTION!  The thin braided wire used in this cable may not be the most resilient to wear&tear

www.ebay.com/itm/Lightning-Line-Out-Dock-to-MICRO-USB-cable-for-hugo-mojo-AMD-n5-iphone-5-5S-6-6S/321954079094
 
Taobao iPhone654-HugoPHA-mojo-lightning-kit otg ios9 (circumvents Apple CCK)

Above image credit: AudioBear
 

world.taobao.com/item/44240667193.htm
 
Fiio L19 (circumvents Apple CCK)
 
Note: some people feel this cable may not be the most reliable choice
 

penonaudio.com/L19-Lightning-to-Micro-USB
 
 
Quote:
The Fiio L19 isn't a Fiio product, I've had this confirmed by Fiio direct. Although I don't believe there are any issues with the cable.

 
Video of Fiio L19 cable working with iOS 9.2 and Mojo
 
 
 
Penon Audio  Lightning to Micro USB Hugo/Mojo/PHA DAC Audiophile Pure Silver Decoding Cable (circumvents Apple CCK)
 
This cable seems to be a popular & reliable choice, BUT PLEASE SPECIFY RIGHT-ANGLED PLUGS, to reduce strain on Mojo & iDevice sockets


 
Right-Angled version:

 
Available from:
 
http://penonaudio.com/Lightning-Pure-Silver-Decoding-Cable
 
 
Zee's Music braided with 8 cores OCC + gold plated connectors & WBT 4% silver solder (circumvents Apple CCK)

Available from:
www.ebay.com/itm/252476440532
 
 
 
We at chord looked at those adaptors but decided they were very likely to potentially put a lot of sheer force right into the most fragile part of to USB Connectors in our unit. this is why we made our adaptor substantially larger to accommodate tour points of connection into Mojo and a cupped end to ensure damaging forces are less likely to be transferred into any single USB socket.

 

 
 
 
  1. Also of interest: www.head-fi.org/t/784602/chord-mojo-the-official-thread-please-read-the-3rd-post/10860#post_12323045
 
 
Please note: Chord Electronics does not specifically endorse any cables which seek to circumvent the CCK, so cannot be held responsible for any issues arising therefrom.
 
 

 
  1. If you have problems with intermittent connection, read this and please note that even genuine Apple CCK lightning cables can sometimes be faulty
 
 
 

Mojo owners using iOS will need to use a software app, in order to output Hi-Res audio through the Lightning connection. There are a few options:
 
  1. Onkyo HF Player ($10 for HD Version)
 
  1. HiBy Music (free)
 

 
  1. Physical attachment without obscuring screen: www.head-fi.org/t/784602/chord-mojo-the-official-thread/2940#post_12033020
 

 

(NB: please also view the VIDEOS section!)
 
IMPORTANT : It is YOUR responsibility to make sure that the cable you are buying correctly fits the connectors on your equipment. None of these links are official endorsements.
 
 
http://www.amazon.com/gp/product/B00FF086HE
 
Sony WMC-NWH10 USB Conversion Cable for Hi Res Audio Output
 

Image credit: Whitigir
 
 
Those who need a cable for digital from zx2/Sony Walkman and can deal with Chinese:

http://m.intl.taobao.com/detail/detail.html?spm=a1z5f.7632060.0.0&id=45034655500

I got one and been using it
biggrin.gif
. Cheap too

 
 
 
  Sony Walkman to HUGO/ Mojo usb connection cable alternative.
 
.... just wanted to share some info which some may find useful if using a Sony Walkman with the Hugo as if like me you have got annoyed at using the special Sony WMC-NWH10 USB conversion cable where you then have to still use a normal usb cable into this bulky not best quality adaptor Sony have ever done which makes it very messy and cumbersome and untidy when strapping the Sony ZX1/ ZX2 to the Hugo. 
  I had already tried a aftermarket cable by Music heaven .... Good news is he has managed to get the Sony NWH10 plug to work with a cable with micro usb on the end to work with his Sony ZX2 and ALO successfully and is willing to make some up for anyone who needs one.  
 
Anyone who is interested just PM Wfanning1 and have a chat with him. 
 
Here are some pics he sent me only the other day of the cable which he is classing as prototypes as he said he will be able to put different cables on with either straight or single ended usb: 
 

 

 

 
 
 
 
 
Also see:
 
www.head-fi.org/t/784602/chord-mojo-the-official-thread-please-read-the-3rd-post/10845#post_12322147
 
 
 
  This cable looks really interesting also!
 
http://www.amazon.com/Custom-Walkman-Digital-WM-Port-Degrees/dp/B00YWEHSQY/ref=sr_1_2?ie=UTF8&qid=1445462623&sr=8-2&keywords=WMC-NWH10

This will not work with HUGO/Mojo. It works only with OPPO HA-2. There's a 20K Ohm resistor between pins 4 and 5.
  .... I did try that cable from Amazon. For this cable to work with Oppo, they have included a 20K Ohm resistor between pins-4 and 5 on the micro-usb side.
I remove the resistor and shorted pins-4 and 5, to see if it will work with HUGO. It didn't.
 

   
 
 
 
We at chord looked at those adaptors but decided they were very likely to potentially put a lot of sheer force right into the most fragile part of to USB Connectors in our unit. this is why we made our adaptor substantially larger to accommodate tour points of connection into Mojo and a cupped end to ensure damaging forces are less likely to be transferred into any single USB socket.

 

 
 
 

(NB: please also view the VIDEOS section!)
 
IMPORTANT-1 : It is YOUR responsibility to make sure that the cable you are buying correctly fits the connectors on your equipment. None of these links are official endorsements.
 
 
 
IMPORTANT-2:  PLEASE CHECK that your Android smartphone supports USB OTG output!! Not all Android phones have this functionality properly implemented. A good place to start, would be here:
 
www.extreamsd.com/index.php/2015-07-22-12-01-14/usb-audio-driver
www.head-fi.org/t/595071/android-phones-and-usb-dacs
 
 
 
IMPORTANT-3:  Please note that you need an OTG cable, not just a standard microUSB cable (although they may look identical, the pin wiring differs):
 
  A USB OTG cable (with a "ID pin 4-connected to-pin 5" micro USB plug) is needed to connect an Android device to a standard USB DAC.
The dual-role Android device is configured to USB host mode (able to interwork with a USB peripheral like Mojo) on the insertion of a USB OTG plug
 
More details can be found at:
www.head-fi.org/t/595071/android-phones-and-usb-dacs
Android USB audio
. FAQ
. A list of USB OTG cables / adapters

 
 
Economical 7cm-long Male-MicroUSB-B-to-Male-MicroUSB-B OTG cable (link)

Please note: although these cheap generic Chinese cables can work OK, the connector quality may sometimes cause an unreliable connection!
 
 
Economical 10cm-long Male-MicroUSB-B-to-Male-MicroUSB-B OTG cable
Available from eBay or aliexpress

Please note: although these cheap generic Chinese cables can work OK, the connector quality may sometimes cause an unreliable connection!

 
ToddTheVinylJunkie (TTVJ) short Male-MicroUSB-B-to-Male-MicroUSB-B OTG cable (Link)

 

 
PenonAudio MICRO USBOTG Silver (Link)


Image credit: sonickarma
 
NOTE!: PenonAudio now offer a Right-Angled plug option, which will be less stresseful for Mojos input socket

 
Available from: http://penonaudio.com/OTG-Pure-Silver-Cable
 
 
 
 
 
 
 
 
                     (image credit: 'Hawaibadboy')
 
 
Moon-Audio (USA) offer premium 7.5cm-long Male-MicroUSB-B-to-Male-MicroUSB-B OTG cable (Link)
 
Custom-Cable (UK) offer premium 25cm-long Male-MicroUSB-B-to-Male-MicroUSB-B OTG cable (Link)


 
Do you need Type-C to microUSB?
 
Monoprice Select Series 2.0 USB-C to Micro B Cable, 6-inch

www.monoprice.com/mobile/product/details/13013
 
 

https://item.taobao.com/item.htm?spm=a1z09.2.0.0.qO1xhm&id=537999317212&_u=s11o9rqq6d35 
 
 

https://item.taobao.com/item.htm?spm=a1z09.2.0.0.JuZF4I&id=529538146962&_u=s11o9rqq4e5b
 
 
 
NOTE: Always be sure that you are using your OTG cable the correct way around! (OTG plug at the phone end, not at the Mojo end)
 
 

 
 
Mojo owners using Android smartphones will need to use a software app to bypass Androids automatic 24/192 upsampling (please also see the 'Informative posts by Rob Watts' section, higher-up this post).
 
 
 

Quote:
  If the native USB audio does not work then you need to use one of the third-party USB audio music players (which include their own USB audio function / soft driver).
 
More details can be found at:
http://www.head-fi.org/t/595071/android-phones-and-usb-dacs

 
 
 
 
There are several apps currently available:
 
  1. USB Audio Player Pro (UAPP) (the most popular - approx. $8) (compatibility-list, & very useful overview, on UAPPs homepage)
 
  1. Onkyo HF Player (approx. $8 for Hi-Res version)
 
  1. HiBy Music (free)
 
 
 
 
 
1) Enter UAPP:
 

 
 
2) Select 'Artists' page:
 

 
 
3) Select drop-down menu:
 

 
 
4) Select Tidal:
 


 
 
You should see this screen:
 

...but if the password screen doesn't appear, then just click on the 'human' icon like this:

 

 
Also, remember to set the quality level:
 



Raw screengrabs credit: maxh22
 
 
 
Note 3 has this weird usb connection.So if I get a Note 3 I will have to get a special usb connection to connect to the Mojo ?


 
that is usb 3.0, you can use standard micro usb 2.0 while pairing note 3 with mojo

 

 
 
  I connected my Mojo to my Samsung S7 and opened USB player audio pro...i get the error 'error initalizing usb system'

 
 
Quote:
  fixed it:
 
ran this
 
https://play.google.com/store/apps/details?id=org.tauruslabs.usbhostcheck

 
 
 
  1. Problems getting Samsung Note 4 to play with Mojo?
 
  1. Problems getting Samsung GS3 LTE to play with Mojo?
 
  1. Information on Android ROMs, in relation to USB Audio
 
 
 
 
We at chord looked at those adaptors but decided they were very likely to potentially put a lot of sheer force right into the most fragile part of to USB Connectors in our unit. this is why we made our adaptor substantially larger to accommodate tour points of connection into Mojo and a cupped end to ensure damaging forces are less likely to be transferred into any single USB socket.


 
 
 
 
 
Here's a wildcard some of you may care to look into: www.head-fi.org/t/784602/chord-mojo-the-official-thread-please-read-the-3rd-post/13440#post_12425559
 
NB: Mojo USB input is:
 
  USB is isochronous asynchronous. This means that the FPGA supplies the timing to the source, and incoming USB data is re clocked from the low jitter master clock. So again source jitter is eliminated.
 
Rob

 
 
  The problem with galvanic isolation is that the USB decoder chip and the data transmitter is powered from the mobile phone. There is no point in having 8 hours on Mojo if your mobile phone battery is depleted earlier. That said, you really need isolation when using a PC; its an order of magnitude smaller problem when using mobile phones, as everything is battery with no ground loops, and mobiles are incredibly power efficient compared to PC's - which gives you very much lower RF noise.

Quote:
  To eliminate the RF & signal correlated noise on USB you need galvanic isolation. The downside to galvanic isolation is that it draws power from the source - which is not something we can do with a mobile product. All Chord desktop DAC's have USB galvanic isolation now.
 
That said, mobile sources are much lower noise - they have very efficient processors, unlike a PC, and there is no ground, so circulating currents are much less, so it is a much smaller problem with mobile. If you can do it, use the optical, as this usually sounds the best and is completely isolated. Optical has a undeservedly poor reputation, as it sounds much smoother and darker than other inputs, and this is just a feature of lower noise floor modulation - its smoother with better instrument separation and focus - but lack of glare is often confused with a lack of detail resolution. Listening tests must be done with a lot of care, as it is easy to draw the wrong conclusions!
 
Rob

 
  .... Mojo has the same USB as Dave - but with Dave it is galvanically isolated, so RF noise and signal correlated currents from the source can't upset Dave at all. I can't do this with Mojo as it draws too much power from the device connected to the USB - but the upside is that mobile sources create much less noise as they are power efficient and there is no ground loops either due to battery operation.
 
So Mojo like Dave has solved the jitter problem via USB, as the timing for the data comes from Mojo not the source.
 
Rob

 
 

(NB: please also view the videos section!)
 
IMPORTANT : It is YOUR responsibility to make sure that the cable you are buying correctly fits the connectors on your equipment. None of these links are official endorsements.
 
 
www.head-fi.org/t/784602/chord-mojo-the-official-thread/2940#post_12033270
 
www.head-fi.org/t/784602/chord-mojo-the-official-thread/2595#post_12029029
 
www.head-fi.org/t/784602/chord-mojo-the-official-thread-please-read-the-3rd-post/12810#post_12407306
 
www.head-fi.org/t/784602/chord-mojo-the-official-thread/3570#post_12046140
 
www.head-fi.org/t/784602/chord-mojo-the-official-thread-read-the-first-post-for-updated-info/4020#post_12054119
 
www.head-fi.org/t/784602/chord-mojo-the-official-thread-read-the-first-post-for-updated-info/4275#post_12057847
 
Please be cautious about head-fier derGabe selling cables  - some members have not received their orders!
 
www.head-fi.org/t/784602/chord-mojo-the-official-thread-please-read-the-3rd-post/11490#post_12349732
 
 
 
iBasso DX90 issues:  www.head-fi.org/t/784602/chord-mojo-the-official-thread-read-the-first-post-for-updated-info/4800#post_12069924
 
 

 
Fiio x3ii and X5ii  owners, please additionally note: www.head-fi.org/t/784602/chord-mojo-the-official-thread-please-read-the-3rd-post/14070#post_12442957
MoonAudio offer suitable cables
Custom-Cable also offer suitable cables

 
[spoiler='Uranus' Co-Axial cable for Fiio X3ii and X5ii] 'Uranus' Co-Axial cable for Fiio X3ii and X5ii

(image credit: 'noobandroid')
 
NB: the DAC end of noobandroid's cable (RH, in the pic) has one too many poles, but, if wired appropriately, may still work correctly (please see the 2nd connection image immediately below). tkteo's Dyson version is what one would actually expect.
 
 
Here are the signal paths for an appropriate Co-Axial cable to connect Fiio X3ii and X5ii DAPs to Mojos Co-Axial digital input:
 

(pin-out identities based upon these: www.head-fi.org/t/784602/14985#post_12467535 )
 
IMPORTANT: James (CEO of Fiio) has privately confirmed to me that the above diagram is definitely correct for X3ii, X5ii, and X7
 
 
Alternatively, if you wish to use a stereo TRS plug at the Mojo end (instead of the mono TS plug in the above picture, which is really all that is required), then the pin connections would be as follows:
 

(pin-out identities based upon this: www.head-fi.org/t/784602/18675#post_12651727)
 
 
  Lately, I have been recommending a certain type of TRS Connector(or Plug) for the coaxial cable to connect a Fiio X3ii/X5ii/X7 to a Mojo, and it appears to be causing some confusion here. Hopefully this post will help clarify this confusion.
 
PLEASE NOTE: This post only applies to Fiio X3 2nd Gen, X5 2nd Gen and X7 devices. Before we get into the topic, here is a picture to understand what TS, TRS, TRRS Connectors are:
 
 
Fiio X3ii, X5ii and X7 devices use a single 3.5mm port that is shared for both Line-Out and Coaxial Digital-Out purposes. A regular coax cable will NOT work with these devices. You need a modified cable with 3.5mm TRRS Connector to work with these devices. The TRRS pins on this coaxial cable have to be configured as shown in the picture below:
 
 ​
 
Now, the coax port on the Mojo is a simple, dedicated 3.5mm coax port. So all it needs is a 3.5mm TS Connector. So in order to connect a Fiio X3ii/X5ii/X7 to a Mojo, you need a coax cable with 3.5mm TRRS Connector on one end and, a 3.5mm TS Connector on the other. The TRRS end goes into the Fiio devices and the TS end goes into the Mojo.
 
EDITOR'S note: There are cable makers who sell this modified cable in both Straight Connectors and Right-Angle Connectors version, but it can be necessary to make a special-order. Dyson Audio used to make them, but this maker became unreliable, and is best avoided until further notice.
 ​
 
If you want a cable with Straight Connectors on both end, you can stop right here and can right away purchase it from one of these places:
Moon-Audio (US) / Custom-Cable (UK) / Uranus (Malaysia)
 
Now the PROBLEM arises, when you want this cable with Right-Angle Connectors. For some reason, the only Right-Angle version of TS Connector, that seems to be available in the market is this plastic one:
 
 
I did not like this plastic connector. I looked around for a solution and I found out that, cable makers like Moon-Audio and Uranus, used a TRS Connector instead on the TS Connector, in their Right-Angle version of the coax cable. But now I had ANOTHER PROBLEM. Uranus-Cable does not ship outside Malaysia and the cable from Moon-Audio was too expensive.
 
  Hope this helps!
 
-EagleWings

 
 
 

(NB: please also view the videos section!)
 
IMPORTANT : It is YOUR responsibility to make sure that the cable you are buying correctly fits the connectors on your equipment. None of these links are official endorsements.
 
  1. Mojos optical input is a standard Toslink Optical socket. However, many DAPs have optical outputs using 3.5mm sockets, so PLEASE CHECK before buying an optical cable
 

 
 
Sysconcept custom-made low-profile optical cable
 
www.head-fi.org/t/784602/chord-mojo-the-official-thread/2370#post_12025183
 
http://www.sysconcept.ca/product_info.php?products_id=349
 

 
 

 
 
Google Chromecast Audio optical cable (6 inches)
 
https://store.google.com/product/optical_cable_chromecast_audio
 

 
 
 
 
 
www.head-fi.org/t/784602/chord-mojo-the-official-thread/2880#post_12032024
 
www.head-fi.org/t/784602/chord-mojo-the-official-thread/2265#post_12023214  (specifically to a Mac computer)
 
related (non-Mojo-specific) discussion about optical cables: www.head-fi.org/t/784602/chord-mojo-the-official-thread/2250#post_12023147
 
[/spoiler]
 

 
www.head-fi.org/t/784602/chord-mojo-the-official-thread/2025#post_12020136
 
www.head-fi.org/t/784602/chord-mojo-the-official-thread-read-the-first-post-for-updated-info/4770#post_12069520
 
www.head-fi.org/t/784602/chord-mojo-the-official-thread-please-read-the-3rd-post/12360#post_12394268
 
www.head-fi.org/t/784602/chord-mojo-the-official-thread-please-read-the-3rd-post/12855#post_12408223
 
....  
And BTW to clear up the Windows Phone confusions, it's confirmed otg is only supported on newer models that supports 'Continuum' for example the Lumia 950/xl. I am not sure about older models but my Lumia 830(running the latest windows 10m preview) does not work with the mojo.

http://forums.windowscentral.com/microsoft-lumia-950/395360-external-audio-dac-support-usb-otg.html

 
 

 
Blackberry Phone
 

 
www.head-fi.org/t/784602/chord-mojo-the-official-thread-read-the-first-post-for-updated-info/5070#post_12078284
 
 
Almost any DAP with a digital-output should function correctly with Mojo.
Some that Head-fiers have successfully used with Mojo include:
 
Fiio X3ii
Fio X5ii
Fiio X7
AK70
AK100
AK120
AK240
AK320
AK380
Shanling M1
Soundaware M1 Esther
 
Also see: www.head-fi.org/t/784602/chord-mojo-dac-amp-faq-in-3rd-post/19620#post_12706423
 

 
 

 
 
ALSO SEE (regardless of which playback software is used): www.head-fi.org/t/784602/chord-mojo-the-official-thread-please-read-the-3rd-post/9780#post_12277215
 

 
 
 
 
  1. Silicone wristbands
  2. Large O-rings from an automotive spares supplier (or these, although they may not last very long)
  3. Sticky back velcro (may leave sticky residue on devices, in the event of trying to remove it entirely)
  4. 3M Dual Lock Low Profile
  5. Double-sided self-adhesive foam (may leave sticky residue on devices, in the event of wishing to remove it entirely) If you use single-sided, you'll still need silicone bands as well
  6. Blu-Tack
  7. 3M 'Command' picture-hanging strips
  8. Cured silicone (just like single-sided foam, this will need silicone bands, as well)
  9. Thin bead of silicone sealant (warning: don't use this method unless you are familiar with how silicone can be removed correctly)
 
 
 
 
or an aluminium case
 
 

 
 
 
 ​
 ​
Chord Mojo + AK70 Music Player Leather Case by Miter
$99.00
 ​
Free U.S. Ground Shipping
 ​
 
 
 
 
 
 
 ​
 ​
Free U.S. Ground Shipping
 ​
 ​
$99.00 Chord Mojo + AK70 Music Player Leather Case by Miter
 
The Chord Mojo + AK70 Music Player Leather Case by Miter made in South Korea, for protecting your Astell & Kern AK70 and Chord Mojo from scratches, shocks and fingerprints.

  1. Hand Crafted MITER Leather Case for Chord Mojo + Astell & Kern AK70 Music Player
  2. Patented / Steel Frame Stand & Case Cover
  3. Protects from Shocks, Fingerprints and Scratches
  4. Hand crafted with Superior craftsmanship
  5. Made in South Korea
  6. Made with “Oil Pull-up” leather, a favorite of Miter, due to its beautiful matte finish, unique in its characteristics with respect to the oily finish different from other varieties of leathers.

 

 

Thank You.​
Drew Baird
Moon Audio 
106 Brady Court
Cary, NC 27511 
919-649-5018 
Drew@moon-audio.com
 
HeadFiLogoSignature.jpg
Follow us Today!​
     

 
 
 
 
Chord have released a high-quality case of their own (please see 'Official Chord Accessories for Mojo', in next section)
 
   
 
Dignis has come up with some cases for the AK70/Mojo combo which looks rather nice.

Photos taken from Dignis Japan's Twitter page.




 
 
 
 
 
Here is a way to seal unused ports on Mojo
 

 
 

 
 



 
 
More pics: www.head-fi.org/t/784602/chord-mojo-the-official-thread-please-read-the-3rd-post/14715#post_12459171
 
Chord case review: https://www.youtube.com/watch?v=J_VuZJH5yvU
 
 
 
 


 
The BASIC USB Adapter Module, that is included in the official Cable Accessory Pack (as shown in the above contents), is used as shown in the following image.
 
It houses the connector plug of the digital transmission cable and thereby reduces stress upon that connector plug and Mojos connector socket.
 
When attached to Mojo, the module also makes it easier to stack with a smartphone without obscuring the screen, since the combination of Mojo+module more closely matches the length of a smartphone.

 
 
blog.moon-audio.com/chord-mojo-review/
 




images credit: http://blog.moon-audio.com/chord-mojo-review/
 


www.modernhifi.de/chord-mojo-zubehoer/  


 
images credit: www.modernhifi.de/chord-mojo-zubehoer/
 
 
 
NB: please also see the Munkonggadgets interview with John Franks
 
 
 
 
 
 
  @Mojo ideas Is there any chance of you making a case that includes the Mojo with the add on module?

 
Mojo Ideas replied to me when I asked if there would be a case to encompass Mojo and SD-card module. Answer, "Yes".

 
 
There was a bit of non-official discussion on this, a year ago - Chord may have different plans now (perhaps not so many modules on the drawing board), but I still anticipate an extended case of some description, to see the light of day, in due course:
 
 
 
. This one is option 3 our favourite option one was disregarded earlier


My vote also goes to this design.

I'd like the silver Mojo which seems to be lurking inside as well please.:hugging:

However what happens (to either solution) if someone wants to clip on one of the proposed add-on modules?

Ian

 
 
I don't anticipate that those cases will be particularly expensive - therefore, it might be quite viable to produce standard-sized versions and also extended versions, to simultaneously encompass both the Mojo and an attached add-on module. This could also confer a nice benefit of reinforcing the 'join' between the Mojo and the add-on.

 
 
 

 


 





(Image credit - Drew  @ Moonaudio)





This is just for fun, so excuse the roughness of this photoshopped mock-up, but this is the kind of thing I was imagining:


Already imagined and more but it can't be quite like that as there are four different module to accommodate it's being refined and costed but it's great you guys can get a hint of where we are going with mojo it's more of a system that a product. But please be patient we want to get all this really right and not just okay.

 
 
 
 
 
The following comment, from John Franks, is mid 2016:
 
my thoughts, too....esp for $100

and if that add-on comes out shortly
what happens then?


Because we have the extender module and the more complex modules underway it's was always our plan to offer a well made longer case too.

 

They are currently in manufacture and you've reminded me to chase the guys.

 
 
 
 

 
This will always be a contentious issue (such is the nature of geeky audiophiles
wink_face.gif
). There will always be fans of one connection type or another.
 
Here are some aspects of the discussion:
 
Rob has a preference for optical on Hugo, but USB on Hugo TT. The difference between them is that TT has galvanic isolation.

Mojo and Hugo don't use galvanic isolation as this would suck power out of the mobile device that they are connected to. You would drain the battery in your phone more rapidly if it had galvanic isolation.

TT, being for home use, doesn't care if it drains power from the device that's connected to its "USB HD" input.

The "USB SD" input on TT doesn't have galvanic isolation and is there for phones. It also works on Windows PCs that do not have the Chord driver installed.

 
  The problem with galvanic isolation is that the USB decoder chip and the data transmitter is powered from the mobile phone. There is no point in having 8 hours on Mojo if your mobile phone battery is depleted earlier. That said, you really need isolation when using a PC; its an order of magnitude smaller problem when using mobile phones, as everything is battery with no ground loops, and mobiles are incredibly power efficient compared to PC's - which gives you very much lower RF noise.

 
Quote:
  To eliminate the RF and signal correlated noise on USB you need galvanic isolation. The downside to galvanic isolation is that it draws power from the source - which is not something we can do with a mobile product. All Chord desktop DAC's have USB galvanic isolation now.
 
That said, mobile sources are much lower noise - they have very efficient processors, unlike a PC, and there is no ground, so circulating currents are much less, so it is a much smaller problem with mobile. If you can do it, use the optical, as this usually sounds the best and is completely isolated. Optical has a undeservedly poor reputation, as it sounds much smoother and darker than other inputs, and this is just a feature of lower noise floor modulation - its smoother with better instrument separation and focus - but lack of glare is often confused with a lack of detail resolution. Listening tests must be done with a lot of care, as it is easy to draw the wrong conclusions!
 
Rob

 
 
 
  USB DAC data is not bit perfect by any stretch of imagination. It tries to be bit perfect, but if it fails to be, it is not corrected at any point. Though again, there should not be any ground and sky differences. 

That is not the case with Chord's windows drivers. If faulty data is sent through, then the DAC requests a repeat, and so ensures perfect data transfer.
 
It is possible with all other OS; but having said that, the data failure rate is very low (otherwise DoP would not work).
 
The USB connection making a difference to the sound is not data related - its down to RF and correlated noise (not jitter as this is completely removed too) - take a look at my previous posts if you are interested. 
 
Rob

 
 
I don't think anyone will ever answer that question satisfactorily if you know how the technical aspects of the devices work. The transport is passing a purely digital signal onto the Mojo, just what the original file is. Mojo does all the work. That being said, some people swear they hear a difference in sound depending on the transport. You decide what is most likely:)

The reasons why sources and digital interconnects sound different are well understood - see some of my posts. In a nutshell it is not jitter (all my DACs are source jitter intolerant) but down to RF noise and distorted currents from the source flowing into the DAC's ground plane. The RF noise inter-modulates with the analogue electronics, creating random noise as a by product, which creates noise floor modulation, and that makes it sound brighter or harder. The correlated or distorted currents very subtly add or subtract to small signals, thus changing the fundamental linearity, which in turn mucks up depth perception.
 
But I also agree in that lots of people hear changes that are not there - I for one have never heard any difference with optical cables (assuming all are bit perfect) with my DAC's, but lots of folks claim big differences. Placebo, or listening with your wallet, plays a part here. Then there are cases of people preferring more distortion... Listening tests must be done in a very controlled and careful fashion, particularly if you are trying to design and develop things.
 
Rob

 
 
 
  Digital transmission is based on SPDIF standard which transmits data and clock information as an encoded signal usually using PCM, that information is decoded on the Mojo into data and clock signal so it's important that the encoded information be jittered free and not degraded over short distance.
 
The USB transmission on the other end is a device to device transmission mechanism using an encoding scheme and handshaking mechanism, it is usually stream based so more tolerant to poorer wire as frames are transmitted and decoded from the source to the target device. The target device will reconstruct the data and clock signal from the frame and then feed it to the DAC to be analog reconstructed and eventually band pass filtered to remove any residual high and low frequency signals out of the audio band.I still think you need to keep the USB cable short but it is more tolerant of longer lengths up to a limit.
 
To make a story short, the short USB cable is fine but an analog cable used as a digital one is just a bad idea. Again, that's just my opinion.


Just to clarify:
 
1. SPDIF decoding is all digital within the FPGA. The FPGA uses a digital phase lock loop (DPLL) and a tiny buffer. This re-clocks the data and eliminates the incoming jitter from the source. This system took 6 years to perfect, and means that the sound quality defects from source jitter is eliminated. How do I know that? Measurements - 2 uS of jitter has no affect whatsoever on measurements (and I can resolve noise floor at -180dB with my APX555) and sound quality tests against RAM buffer systems revealed no significant difference. You can (almost) use a piece of damp string and the source jitter will be eliminated.
 
2. USB is isochronous asynchronous. This means that the FPGA supplies the timing to the source, and incoming USB data is re clocked from the low jitter master clock. So again source jitter is eliminated.
 
So does this mean that any digital cable will do?
 
Sadly no. Mojo is a DAC, that means its an analogue component, and all analogue components are sensitive to RF noise and signal correlated in-band noise, so the RF character of the electrical cables can have an influence. What happens is random RF noise gets into the analogue electronics, creating intermodulation distortion with the wanted audio signal. The result of this is noise floor modulation. Now the brain is incredibly sensitive to noise floor modulation, and perceives this has a hardness to the sound - easily confused as better detail resolution as it sounds brighter. Reduce RF noise, and it will sound darker and smoother. The second source is distorted in band noise, and this mixes with the wanted signal (crosstalk source) and subtly alters the levels of small signals - this in turn degrades the perception of sound stage depth. This is another source of error for which the brain is astonishingly sensitive too. The distorted in band noise comes from the DAP, phone or PC internal electronics processing the digital data, with the maximum noise coming as the signal crosses through zero - all digital data going from all zeroes to all ones. Fortunately mobile electronics are power frugal and create less RF and signal correlated noise than PC's. Note that optical connection does not have any of these problems, and is my preferred connection. 
 
Does this mean that high end cables are better? Sadly not necessarily. What one needs is good RF characteristics, and some expensive cables are RF poor. Also note that if it sounds brighter its worse, as noise floor modulation is spicing up the sound (its the MSG of sound). So be careful when listening and if its brighter its superficially more impressive but in the long term musically worse. At the end of the day, its musicality only that counts, not how impressive it sounds.         
 
Rob

 
 
 
Two questions related to Rob Watts' comments on optical output as a source to the Mojo;

1. What is audio "glare"? What does it sound like? And how does one distinguish it from detail?

2. Is the conclusion that one with an Apple computer should be using its optical/toslink output (that also serves as a headphone output) rather than its USB output, even if one uses an AudioQuest Jitterbug, Schiit Wyrd, Uptone Regen and an Akiko USB tuning stick, or similar devices ?

Glare is normally used for extreme form of hardness or grain in the treble. So I guess one could say going from bad to good glare, grainy, hard, bright, smooth, dark. 
 
Distinguishing it from detail is tricky as a brighter sound is easy to confuse it with more detail resolution. Indeed, truly more transparency, does sound brighter. So you have to be very careful, and I have been caught out in the past. One way of recognising it is with timbre - it the extra brightness is noise floor modulation for example, then all instruments will sound brighter - even those that are supposed to sound rich and dark. But if the brightness is better detail resolution, then smooth instruments will just sound clearer, not brighter. Also, if instrument separation and focus is worse, then it is not more transparency.
 
When somebody says it sounds better, but can't actually describe in details what the differences are, be warned! They may be preferring distortion. Fortunately, our lizard brain ignores all this - if its really better, it will be more emotional and involving, so you should use this as your goal. But assessing whether its more emotional or musical takes a lot of time, you can't do it on a quick AB test.
 
The USB filter devices help (hopefully) but do not solve the problem. It has to use galvanic isolation to do it properly.
 
Rob

 
There are pros and cons to each connection type. Optical may be the most immune to RF, but most people encounter no RF issues with USB or Co-ax, so it is best to choose your connection type based upon what is offered by your existing player, and then to learn how to get the most reliable result from that connection type. Mojo itself has the potential to get equally-superb SQ from Optical, Co-ax, or USB.
 
  Just to make it 100% clear - the USB input will measure absolutely identically to the coax or optical inputs if the USB data is bit perfect.
 
I have set up my APX555 so that it uses the USB via ASIO drivers, and I get exactly the same measurements on all inputs - 125 dB DR, THD and noise of 0.00017% 3v 1k 300 ohms. I have done careful jitter analysis, FFT analysis down to Mojo's -175dB noise floor, and can measure no difference whatsoever on all inputs (with the APX always grounded on the coax).
 
If somebody does measure a difference its down to mangled data on the USB interface (or perhaps poor measuring equipment - Mojo is way better than most test equipment). Mojo can't convert 16 bit data back to 24 bit....
 
Rob 

 
 
If you're still wondering about the very small percentage of RF issues, then please skip down 6 sections below
 

 
Android automatically upsamples ALL music files to 24/192, which is not a good thing for Mojo - here's what to do about it (IMPORTANT!: please see discussion in the 'Connecting Mojo to OTG (microUSB) devices' section, above)
 

 
1) Enter UAPP:
 

 
 
2) Select 'Artists' page:
 

 
 
3) Select drop-down menu:
 

 
 
4) Select Tidal:
 


 
 
You should see this screen:
 

...but if the password screen doesn't appear, then just click on the 'human' icon like this:

 

 
Also, remember to set the quality-level:
 



Raw screengrabs credit: maxh22
 

 
www.head-fi.org/t/784602/chord-mojo-the-official-thread/195#post_11994409
 
www.head-fi.org/t/784602/chord-mojo-the-official-thread-read-the-first-post-for-updated-info/4200#post_12056432
 
www.head-fi.org/t/784602/chord-mojo-the-official-thread-read-the-first-post-for-updated-info/4725#post_12068719
 
www.head-fi.org/t/784602/chord-mojo-the-official-thread-please-read-the-3rd-post/8340#post_12217948
 
www.head-fi.org/t/784602/chord-mojo-the-official-thread-please-read-the-3rd-post/9750#post_12274956
 
Thanks to 'DanBa' for this informative post:  www.head-fi.org/t/595071/android-phones-and-usb-dacs/7350#post_12061822
 
 
Also consider:
 
www.head-fi.org/t/784602/chord-mojo-the-official-thread-please-read-the-3rd-post/13725#post_12433418
 
www.head-fi.org/t/784602/chord-mojo-the-official-thread-please-read-the-3rd-post/13800#post_12435426
 
www.head-fi.org/t/784602/chord-mojo-the-official-thread-please-read-the-3rd-post/13575#post_12428810
 
www.head-fi.org/t/784602/chord-mojo-the-official-thread/2280#post_12023830
 
www.head-fi.org/t/784602/chord-mojo-the-official-thread/2910#post_12032759
 
www.head-fi.org/t/784602/chord-mojo-the-official-thread-please-read-the-3rd-post/14775#post_12460426
 
If you're using Neutron & can't output higher than 44.1khz
 
If you are experiencing clicking sounds, you may need to adjust buffer settings:
 
www.head-fi.org/t/784602/chord-mojo-the-official-thread-please-read-the-3rd-post/9930#post_12282635
 
www.head-fi.org/t/784602/chord-mojo-the-official-thread-please-read-the-3rd-post/10230#post_12295303
 
and you should also consider if it might be an RF interference issue (see the relevant links, on that issue, below)
 

 
 

 
 
  guys sorry to ask but this volume table is for the hugo...I can't find a similar one for the mojo.
 

 
 
1/found my volume setting is blue to purple-ish (using flc 8s...have a pretty good seal, too)
based on this table i'm a bit concerned.... am i going deaf? (this is when outside, walking around, however)
 
2/also find my battery life is not 8 hrs...more like 4-5 based on my listening so far....anyfind have similar experiences?

The color setting for Hugo and Mojo is the same - the only difference is below -43 dB and above +3dB the differing light scheme kicks in. Also you can see variations in the two balls color as it gets closer to the next level.
 
As for battery life, this depends upon how well charged it is, and whether you are using USB or optical/coax, how loud you play, and the impedance of the headphones. So driving 300 ohms, using optical, green volume, you will get 8 hours. Use USB it will be 6 and a half hours. Use a low impedance IEM, green volume, and it will get worse. Use an 8 ohm loudspeaker and you will get even lower battery life.
 
Rob

 
 
Thanks to GRUMPYOLDGUY for creating the following Mojo-specific spreadsheet:
 
  The color indicator/volume problem isn't really a problem. Just start the Mojo in preset mode and count clicks to get to the right output level.
 
Preset is 3Vrms, each click is 1dB... Here's a summary of the math to get to your desired dB SPL level...
 
You need to know two parameters about your headphones:
 
Nominal sensitivity (dB SPL/mW) 
Nominal impedance (Ohms)
 
 
Step 1: Calculate sensitivity as dB SPL / 1 Vrms
Y = y0 + 10*log10(1000/Z)
y0 = headphone nominal sensitivity (dB SPL/mW), Z = headphone nominal impedance
 
Step 2: Pick a target volume (dB SPL) and figure out output level needed to achieve it
V = 10^((T-Y)/20)
V = target level (Vrms), T = target loudness (dB SPL), Y = calculated sensitivity
 
Step 3: Loss due to Mojo output impedance
Loss(dB) =  20*log10(Z/(Z+0.075))
Z = headphone nominal impedance, 0.075 = Mojo nominal output impedance
 
Step 4: Find how many dB down the target level is from the preset
L = 20*log10(V/3) - Loss(dB)
 
Simply round the number above to figure out how many clicks from the preset you need to go. 
 
I created a little spreadsheet to calculate the right levels for directly driving headphones from the Mojo... 
 

 
 

 
 
 
To set the output level to 3V ( line level ) for connection to a preamplifier press both volume buttons
together when switching on the unit. Both volume balls will illuminate light blue. This mode is not
remembered so when you switch off it will reset back to the previous volume stored for safety reasons.
 
 
 
  Has anyone experienced the Mojo keeping its 3V setting even after turning off and on again? It happened twice to me. Fist time, it almost killed my Fitear 335. Luckily I notice the huge hiss at the beginning of the song and unplugged it quickly. Second time, I saw the volume balls was illuminated in violet and had to decrease the volume (usually they are pink-ish).

 
The Mojo remembers what volume you had last and does not reset when you turn it off.


Line out mode is an exception, unless you have pressed a volume button while in it. Then the volume you set will be saved, if you did not change volume in Line out mode, it will not save to 3V for obvious reasons.

 

 
To set the output level to 1.9V RMS, first follow the above guidance, to attain 3V, and then continue further, with the following:
 
  Yes 4 clicks down will set it to 1.9v (both balls indigo). Each step is always a 1 dB change.

 
 
 
Please no worries!
However, iwas wondering if you could answer, the Mojo on line level mode - does this still run thru the Mojo's amp? from how i understand your earlier descriptions, buth the amping and DAC is done in the FPGA?
thus there is no way to truly use it as a dac without double amping?

Line level mode is just a volume preset for the volume control - nothing else changes.
 
Mojo has an FPGA (which is digital logic only) a discrete DAC (turning digital signals to analogue via flip-flops and resistors) and a single output amplifier - and that is it.
 
Conventional DAC headphone amps use differential outputs and have two I to V converters (current to voltage), a differential to single ended converter, and an output amplifier. Wrapped up with that is a analogue filter. So that's a lot of passive components and four amplifiers in the signal path. 
 
Because Mojo's FPGA has extensive digital filtering (at 2048 FS) and has a noise shaper that runs at a very high rate (104MHz) and uses a discrete DAC, I can keep the analogue section radically simpler, and this is one reason why Mojo is so transparent compared to all other DAC amps.
 
Rob

 
 
  @xtr4 i understand the FPGA designs makes the dac and amp essentally the same... what im really trying to get at is, can the FPGA's amp functions be bypassed so it is used simply as a DAC, and the two 3.5mm outs are true line outputs to prevent double amping
Paste

No, you need at least one amplifier to do the critical I to V conversion. Now it is possible to design a voltage only DAC (no amp at all), but they sound poor due to lots of problems - the largest being the huge amount of distortion you get doing it that way. Believe me, if I could make it simpler I would. The key that Mojo has is extremely low distortion and noise (0.00017% 3V 300 Ohms) but only one single amplifier in the signal path - and this amp combines headphone drive, filtering and I to V conversion in a single stage.
 
Rob

 
 
For physically-connecting Mojo to active speakers, see: www.head-fi.org/t/784602/chord-mojo-the-official-thread-please-read-the-3rd-post/13350#post_12423100
 

 
 
  I just carried a quick amp test and my results are as follows:
 
Portaphile 627x: Sadly this amp proves to be the worst offender. Mojo has out classed this amp by a large margin. 
Meir Audio quickstep: This amp did not alter the sound but to me there is no point in pairing it with quickstep as Mojo alone offers far more volume than paired with quickstep.
Wagnus Epsilon S: Expanded the soundstage which was nice but like the other amp the amp section is just nowhere as powerful as Mojo. I felt transparency also took a hit.
Analg2paper TR-07hp: This was the best pairing of the lot. Like other amps the transparency took a hit but the added bonus was the bass had a nicer reverb. To my ears the bass become extended the the decay was a lot more natural. The mid-bass to my ears was reduced and sub-bass become a little more prominent. 
 
Summary: Add an amp if you like to color the sound and play around with the tuning, I see no real value in adding amp. So far I dont have any amp that is as powerful as mojo. 

 
 
IMPORTANT: Please see the earlier section, on 'Regarding Mojos Output Stage'
 

 
This post was the stimulus for Rob's responses, quoted below:  
Quote:
 
  It is always better to give Mojo bit perfect files and let Mojo do the work, as the processing within Mojo is much more complex and sophisticated than a mobile or PC.
 
So when you have an app that has a volume control, and no bit perfect setting, then set it to full volume on the app on the assumption that this will keep the data closer to the original file.
 
The volume control function on Mojo is much more sophisticated than the PC as I employ noise shaping and I do the function at a very high internal sample rate. Hopefully using the volume set to max on the app will mean the volume coefficient is 1.0000000... so it will return the original data.
 
Rob 
  You can always do a listening test. If set to max against 50% say, and it hardens up (becomes brighter) with loud recordings then its clipping. If on the other hand the perception of sound-stage depth is reduced, then the volume control is degrading the sound.
 
If you do that test and can hear no difference then don't worry, its a good app volume control.
 
Rob
 
PS for fun I just did a very quick test using Dave. I listen to radio 3 using the BBC iPlayer. I normally have it set to max. I reduced the iPlayer volume control to half, boosted Dave volume control by +6dB - and yes I felt sound-stage depth was worse with lower iPlayer volume.

 
 
IMPORTANT!: Android automatically upsamples ALL music files to 24/192, which is not a good thing for Mojo - here's what to do about it
IMPORTANT!: please also see discussion in the 'Connecting Mojo to OTG (microUSB) devices' section, above)
 
 

 
Please note: Instructions advise charging a brand-new Mojo for 10 hours before using, but actually, it is only necessary to charge until the tiny white charging LED goes out. With most brand-new Mojos, this will be around half that. Just trust what the charging LED tells you.
 
 
For a Mojo that has already been charged previously:
 
Quote:
Originally Posted by Rob Watts 
 
Charging times - its 4 to 5 hours if the unit is off from flashing red to full charge. But if you are using it at the same time, it will take much longer (maybe 12 hours), as current is being drawn to feed Mojo and less to charge the battery. Check that the charging LED is not flashing, as this indicates a fault such as insufficient current from the PSU.
 
Rob

 
 
If your Mojo LED is flashing whilst connected to a charger, please check that the charger is rated at least 1amp current-delivery
 
 
Quote:
 
@Rob Watts

Just a quick question. Ive been using my Mojo just below the double red volume. Got 4hrs out of it and the battery indicator went yellow. Does this seem about right? Feels like i may only get 8hrs from a full charge at well below my normal listening level. Im assuming harder to drive headphones may get 5+hrs out of a charge. This seems lower then i expected.

I have dug out my original design notes and measurements from the battery ADC built into the Xilinx FPGA from one of the prototypes.
 
The intended colours for battery life are:
 
Blue              100% to 80%
Green             79% to 50%
Yellow            49% to 10%
Red                  9% to 2%
flashing red     less than 2% or 10 minutes left.
 
Use this as a rough guide only, as the battery voltage and life left was not exact!
 
Rob

 
Quote:
Originally Posted by Rob Watts /img/forum/go_quote.gif
 
.... when battery is fully charged even after 10 hours or 5 led indicator is automatically turn off, when I put it to charge after charging the led indicator turn on again ( white color ) and looks like battery never been charged .


Don't worry that's fine, the charger has been reset and its in trickle charge mode.
The colour without the charger connected is the one to watch (fully charged is blue then green, yellow down to red, flashing red means 10 minutes left).
 
Rob
 
 
First, check that your charger is rated for at least 1amp charging current (higher is fine; lower is not). If the charger is not rated high-enough, then Mojos white charging LED will flash, to warn that Mojo will not charge successfully.
 
However, if your charger is fine, then it may be that Mojos battery has been discharged more-deeply than usual:
 
Quote:
 
  .... I am thinking the battery is not holding any charge. 

Try charging overnight with the unit off. The charging circuit looks at the state of the battery before charging. If the battery has a very low voltage, it will trickle charge the battery until it gets to a safe voltage, and then full charge will commence. This trickle charge mode can take several hours, and it is done for safety reasons, and it will appear that the battery is not working as the trickle charge mode takes some time. When in this mode Mojo must be off.
 
When charging make sure the battery light is white and not flashing - if it flashes, pull out the charging cable, count to ten, re-attach the charging cable. If it continues to flash, it is most likely the charger is not giving 1A at 5V, so use a better charger.
 
Rob 

 
Originally Posted by Rob Watts
 
 
Firstly the 4/5 hours is the charge time whilst it is in constant current or full charge mode - so that will get you to blue. But after that it goes into trickle charge mode, and the white light will still be on. I can't remember how long the trickle charge mode is, but I guess 9 hours would be right for full and trickle charge.
 
Of course, if you are charging whilst on it will take very much longer to charge, and the charger timer might get triggered then you get the flashing white battery LED. The charger timer circuit is only on during full charge mode. So if it's fully charged, and then you plug in the charger and turn it on, then the white charger light will stay on permanently as the trickle charge is being balanced by the current Mojo is drawing (no net current into or out of the battery).
 
Hope that explains!
 
Rob

 
 
 
It can push 5.3V when fully charged.


Charging state of the battery makes little difference to the output level and mojo actually has no problem in driving HD 800's by our precise and accurate measurements. There are no significant or measurable changes to the output with these headphones so possibly it's something else happening here. Interesting though!

 
 
Quote:
 
Originally Posted by Mojo ideas /img/forum/go_quote.gif
Quote:
I am thinking of buying a Mojo (already have a Hugo) as it is more portable than the Hugo and easier to charge (no need for the specific charger)

One question I have for Chord is how many charge/discharge cycles is the Mojo battery going to last ? I would expect 1000 cycles at least before 80 % capacity is used.

Anyway the Mojo is not a mobile phone that needs to be on 24 hours a day anyway.

And is this battery use replaceable in 5 years time?

Thanks

Well in excess of 2000 however that is quoted as full deep discharge so any lesser level of discharge counts towards a full one so you'd need two 50 percent cycles to count as one full one this means that as our battery is unlikely to undergo repetive deep cycles it's life is calculated to be out beyond 12 years . The battery is a brand new advanced design. It can be replaced easily in any-case as it has a plug in connector,
 
Quote:
Originally Posted by Mojo ideas /img/forum/go_quote.gif
.... the batteries are expected to last far longer than three years. The batteries in our designs are not subject to damaging deep discharge cycles or anything more than very light current demand .... Batteries used for power tools are quite a different matter, but in our units expect a life of greater than ten years. Mojo has a plug on it so it's just an easy replacement for a shop technician Hugo batteries will need soldering in place though but this is also a low skilled job which would be require rarely if ever.
John Franks.

Quote:
Originally Posted by Rob Watts /img/forum/go_quote.gif
The batteries are plug in and held into place via thermal sheet. Very easy to change by your dealer.
 
You should see more than 10,000 hours of use before the battery will need changing.
 
Rob
 
We had the battery developed for only our mojo application. Done for us especially, It took Chord 3 years and many attempts to get the sheer ear thumping power density we have achieved in mojo. So I'd rather people didnt underestimate our design skills and I'd ask please don't think you can better it with a quick battery substitution as this can be risky or even dangerous.

 
Quote:
Guys we at Chord really did extensive studies into batteries before we chose the optimised solution we have in Mojo. We even looked at up and coming battery technologies like lithium sulphur which potentially could extend Mojo's playing time by a factor of four but unfortunately the newer chemistry's are just not there yet. Be aware the battery technology we've chosen is good and above all its safe being a higher spec than most. Remember that mojo is designed for a pocket or a hand to carry and a poorly deigned lithium ion battery could be chemically very volatile. Look up lithium batteries catching fire I don't want one of those in my pocket!

 
 
 
There is a minute amount of battery usage when the unit is turned off as it has to monitor the button states. So it can be expected that it will need a small charge after not being used and off, this shouldn't be a problem as it will be a few months before mojo loses all charge when switched off.
 
With regards to the 3 buttons flashing, this is not a problem when the unit is charging. 

Mojo will discharge the battery - it will take about 6 months to do. But after finishing a charge, if you reconnect a bit later it will re-charge with the white light on. But it is only supplying a few milli-amps of current, as the last part of the charge is a trickle charge.
 
So the white light on is nothing to worry about.

Originally Posted by Rob Watts

 
the battery is always connected - but - when the charger is on, and the battery is fully charged, then the trickle charge is balanced by the current that the amp needs, so no nett charge going into the battery - its just going from the charger to the amp... The battery is still providing a low impedance, and dynamic surge currents though, but the average DC current is just matched by the charger.
 
Rob 

 
 

 
Is there any harm in most of the time leaving the Mojo hooked up to computer and plugged in to the wall to keep a full charge?

No it's fine to leave it plugged in all the time but if your charging from an unplugged lap top you may drain the lap tops batteries. But if your using just the data USB connection ithe mojo takes no power from the connected device. John F.

 
Quote:
 
Mojo, like Hugo, has been designed so that you can have the charger plugged in constantly. So on a desktop charge and run it at the same time. Once its fully charged, the charger will just supply enough current to balance Mojo's current draw, so no net current from the battery.
 
Rob

 
Quote:
 
Just to clarify. Charging is automatic. If you are playing and charging at the same time, with a fully charged battery, the charger will supply enough current to balance the consumption used by Mojo, so no net current into the battery. If its fully charged and the unit is off, the charger will go off. The charger will re charge automatically when the battery voltage falls to 8.2v (off at 8.4v) so keeping the charger connected will ensure a full charge.
 
Rob 

 
Quote:
 
  I am also considering replacing my desktop DAC (which, by the way, cost far more than than the Mojo) with a second Mojo, so I might have something to sell too! One question that I think remains unanswered is, Is it alright to leave the Mojo on 24/7 plugged in with a 2A wall wart? Will that adversely affect the battery or anything else in the Mojo? Would keeping the Mojo on all the time avoid the issues with the battery charging circuitry?

It was designed to run this way. If you want to maximize battery life, then turn Mojo off when not using it, with the charger connected permanently and it will be fine as the charger will disconnect automatically, and re-charge automatically when the battery voltage drops.
 
Rob

 
 
 
Quote:
 
Hi Rob, and John,

1. If I leave the Mojo plugged in to the wall wart all the time, that should be fine, correct?
2. I presume that any time I want to resume listening in its present setup (meaning listening via the computer and with the charging cable still connected to the wall wart), the Mojo should be "ready to go" and that there is no need for me to ascertain the battery level? 
3. If I then use the Mojo as a portable, and later on connect it back to the computer for listening (with the charging cable then connected to the wall wart again), what would be the minimum color of the battery light? Green?

 
1) yes it should be fine
2) overall there is a slight net drain on the battery so starting at blue or green on the battery indicator is a good idea but not mandatory
3) difficult to answer as it depends how long you've been listening whilst mobile but see answer above. Happy listening George!

 

John answered 1 and 2 fine, but I thought I would clarify exactly what happens when you charge and listen at the same time.
 
I use a dedicated charging chip for the Li battery, and this has a number of safety features, and works with a number of settings to ensure safety.
 
Now one of the safety circuits is a safety timer, and this is when the charger is in full charge mode. This timer is set to about 8 hours, and normally full charge mode takes 4 hours, when the unit is off. But when the unit is on and playing, there is a risk that the safety timer will be set, as it can take 12 hours to fully charge (from flashing red) and when playing music (for those 12 hours) at the same time. If the safety timer is set, then the battery LED will slowly flash white, and no further charging will take place. To reset the timer, just disconnect the charge USB, wait 10 seconds, and reconnect, and it will recommence charging. So if you are charging and playing, then when you have finished listening, turn Mojo off, and it will be OK. When Mojo is blue, and you connect the charger, then it is trickle charge mode, and the safety charger is not operating. So if Mojo is green, the safety timer won't trip out, as it will play and leave full charge mode within 8 hours, so you will be OK. It should be OK at yellow too. I guess the easiest way of dealing with it is to turn Mojo off after listening, then you will be fine, unless you listen for longer than 8 hours starting from fully flat.
 
Note that you can get the flashing battery LED if the USB charger voltage is low, from a charger that can't supply the current, or a USB cable that has high resistance. But you will see this pretty early on.
 
I hope this clarifies.
 
Rob

 
Quote:
Originally Posted by Rob Watts

 
the battery is always connected - but - when the charger is on, and the battery is fully charged, then the trickle charge is balanced by the current that the amp needs, so no nett charge going into the battery - its just going from the charger to the amp... The battery is still providing a low impedance, and dynamic surge currents though, but the average DC current is just matched by the charger.
 
Rob 

 
Quote:
  If you fully charge Mojo then use it in a desktop it will not switch off; the power dissipation that the charger uses in matching the current drawn by Mojo is negligible. You are only at risk when charging and using at red ....
 
Just to give you some numbers - fully charged and matching Mojo's current draw the power dissipation is 107 mW for the charger circuit. That will increase running temperature by less than 1 deg C. But at flashing red it is 910 mW for the power dissipation in the charger.
 
Now I could fix this by using a switcher based charger rather than a linear one - but these inject too much RF noise onto the battery. This would impair sound quality, and Mojo's design goals was that plugging in the charger would have no significant change in SQ - which would not happen if I used a switcher based charger. I am not prepared to damage SQ as to me this is the most important aspect just for a tiny improvement in usability.
 
Rob

  To understand it better, let's assume Mojo is off and charging.
 
Now the charger has two modes of normal operation - constant current, which is set to 330 mA, and constant voltage which is set to 8.200 V. Now when the non charging battery battery voltage is less than 8.200 V, then the charger supplies a constant current. But when the non charging battery voltage gets close to 8.200 V, then the charger switches mode to constant voltage at 8.200 V. The current that is charging the battery then slowly falls from the initial 330mA, to zero - its in the trickle charge mode now. Eventually, the non charging battery voltage hits exactly 8.200 V, the charger is in constant voltage mode of 8.200 V, no current now flows into the battery, and the charger switches off automatically. When the battery voltage falls to 8.0 volts, then the charger will return to charging. Tip - if you want to force the charger to top up Mojo's battery to 8.200 V then removing the charge USB, wait 5 seconds, reattach, and the charger will top it up to 8.200 V.
 
Now imagine that Mojo is on at the same time as it is charging. In this case, the battery will continue to charge until it gets to 8.200 V, and the charger is set to voltage mode and gives 8.200 V too; so no current flows into or out of the battery; but Mojo itself is drawing 180 mA of DC current, and this will simply come from the charger - so the charger will supply the needed 180 mA for Mojo. It will do this for ever, and it won't switch off. This is intended, as it means that the battery is effectively not being used to supply the bulk of the current, won't charge or discharge, is held at a safe level, and will operate like this for a very long time.
 
Now we have been talking about DC currents, and this is indeed the vast bulk of the current. But what about dynamic currents and noise? Because the output impedance of the battery is much lower than the charger, then the noise of the charger is reduced; also dynamic currents still comes from the battery. So running in this mode ensures the best of both worlds - low RF and audio band noise from the battery, large dynamic currents available, and low PSU impedance too - but without the worry of the battery wearing out from charge and discharge cycles.
 
I hope this clarifies.
 
Rob 

 
Quote:
when the Mojo is being used and being charged especially when driving lower impedances there is a net drain on the battery. This means that the charging circuit does not quite provide enough power to power the Dac and amp circuitry and keep charge the battery at the same level this is because we had to limit the amount of charge over a given time due to thermal constraints. Our charging time with the Mojo switched off is usually four or max five hours this is a little inconvenient but when we compare this to other Dac amps that need up to a full twenty four hours to charge we feel that we didn't do such a bad job.

 
 
 
 
Broadly-speaking, most people get around 7-8 hrs from a fully-charged Mojo, but it can vary depending on, for example, what load your IEMs/CIEMs/Headphones present to Mojos output stage, how loudly you play your music, and also (to a small degree) what digital protocol you are using:
 
low load, opt/coax may yield closer to 8 hours
low load, USB may yield closer to 7 hours
 
heavily loaded then you could lose another hour.
 
Low load would be -20dB FS into 300 ohms, 3v preset volume.
 
 
 
Quote:
  As for battery life, this depends upon how well charged it is, and whether you are using USB or optical/coax, how loud you play, and the impedance of the headphones. So driving 300 ohms, using optical, green volume, you will get 8 hours. Use USB it will be 6 and a half hours. Use a low impedance IEM, green volume, and it will get worse. Use an 8 ohm loudspeaker and you will get even lower battery life.
 
Rob

 
Quote:
   
 optical is the lowest power - the USB decoder chip is about 1/3 W and is turned off when VBUS is low.
 
Rob
 
Quote:
  .... the impedance of headphones can be reactive, so there could be a phase shift between current and voltage, thus slightly increasing power drain. But don't worry about that. The max power being drawn within Mojo is with 8 ohm IEM's - as although the power in the load is low, the current is higher and that will increase power dissipation on Mojo's discrete OP stage, even when you are running say at red on the volume.
 
Oh another point - power loss within the battery is 0.2% of Mojo's total, so it is insignificant.
 
And another one - power loss whilst charging is because of a use of a linear charger - so when the battery is fully depleted, we get max power loss in the charger, and very little power loss whilst at the end of the charge cycle. Why do I use a linear charger and not a switcher based charger? I have yet to find a switcher charger that allows full RF filtering, so it will upset the sound quality whilst charging. My design goals were to have no loss in sound quality whilst listening and charging and going to the current range of switcher chargers won't do that.
 
Should you find the thermal trip operating whilst charging and listening, then as a poster recommended, putting Mojo on its side fixes that possibility.
 
Rob

 
 
 
Originally Posted by Rob Watts

Does it do any damage if one inadvertently puts power into the micro USB data port?
 
Not if it is a legal VBUS +5v. Now Mojo is protected for over-voltage on VBUS, to prevent possible failure of the battery charger; this protection will fail if too much voltage and current were on VBUS; if the protection diode fails it will short, thus ensuring that dangerous over-voltage will never damage the battery charger; a failed protection diode would then need to be replaced. That said, I am not aware of any failure of protection diodes at all.
 
Rob 

 
 
 
Quote:
 
 
  I found the reverse.  I'm using Sennheiser HD-25 1 II: directly out of the Mojo the sound seems present and correct, but when used with a Ray Samuels SR-71a, the sound goes to a whole new level.  The sound becomes rock solid and more like listening to musicians playing instruments; without the Ray Samuels the sound seem to collapse in on itself and become more hi-fi (ie impressive noises but less music).  To my ears, the extra amplification is not adding tonal euphony but is instead making the most of the DAC.
 
I have a theory that it's to do with the power supply: when using headphones more current is drawn and in a varying manner, ie it varies with the music.  This varying of current affects (I think modulates) the power supply voltage which affects the DAC, amplification and ultimately the sound. By connecting directly to an amp, there is less current drawn and no variation.  This might also explain why companies such as Naim claim improvements to their amps' sound quality when external power supplies are added.  Just my 2CW.


I do not buy this all. You need to bear in mind several facts:
 
1. The battery is capable of delivering 3A of current, and has very low impedance.
2. Mojo amplifier has a very high power supply rejection ratio.
3. The output is pure class A at 5v RMS into 300 ohm.
4. Reducing the output load only starts to increase distortion with 33 ohms - at this level it is very much lower than other headphone amps. The HD25 is a very easy 70 ohms.
5. Mojo is designed to drive loudspeakers. You will be amazed hearing it fill the room with beautiful sound using efficient 8 ohm horn loudspeakers.
 
Some people like the sound of more distortion - 2nd harmonic fattens the sound making everything sound phat, soft and rounded. But its not natural, nor do I find it musical, as everything sounds phat. I want soft sounds to sound soft, and sharp sounds to sound sharp - not everything to have a soft sheen on things all the time.
 
I can give you another example. I just had an email today from a very experienced dealer that asked me this question:
 
"Chord Mojo should have single amplification which drive to both headphone output, but if I'm using any headphones (HD800 for example, very heavy to drive) to put on headphone output 1 and I connect another headphone to headphone output 2 (Beyer T1 for example), I hear no differences on sound quality. Normally if one amplification section used to drive 2 headphones output, then once we connect the second headphone will make overall sound quality degradation (as the case of Beyerdynamic A20 amp, Grace Design M903, etc). What is the logical explanation of this? As it seems Mojo has 2 separate amplification sections which drive independently for each headphone output."
 
Of course Mojo does not have two amplification stages. It can drive two headphones with ease because it has exceptional low output impedance, and it has exceptional current linearity. So loading it with more headphones has no effect, unlike other headphone amplifiers.
 
Indeed, when I initially started designing Hugo I was shocked how poor from a measurement point of view headphone amps were. Poor output impedance, huge levels of distortion, and poor current linearity seem to be typical. And these things matter, if your goal is transparency and musicality. 
 
Rob
 
Quote:
  The earlier prototypes had less current than Hugo - 0.2A RMS. But last Feb we were with Nelson (Malaysian distributor) and I showed him the prototype. He brought some headphones, and we compared it to Hugo - and it was bad, a lot of clipping from Mojo, none from Hugo. It so happened (luckily) Nelson gave us possibly the only 8 ohm headphones on the market Final Audio Pandora.
 
So I upgraded the current to match Hugo, its the same 0.5A RMS. To do this I used 6 small OP transistors in parallel as size would not permit use of the large devices I use in Hugo.
 
Mojo development had some weird fortuitous events - the first time we showed the prototype happened to use the only 8 ohm headphones that would show problems - and the day we were deciding on the design Xilinx emailed me with a new FPGA that would enable Hugo performance but with the needed power, and happened to be in production just in time for Mojo. 
 
 
  Does anybody know the battery capacity on the Mojo? sometimes i use a 5000mah external battery, which the Mojo completely drains, and it barely goes to green level(i get about 4--5 hours of use after this charge). Also, i read that Chord recommends charging from a 1A/5V source. Is there any problem if i charge with a 2A/5V iPad charger? Honestly i didn't check to see if it shortens the charging time. 
The thing is i travel a lot, and i prefer to have with me a single 2A charger, which i can use for my iPhone, iPad, and hopefully for the Mojo.

Mojo's battery is actually about 14Wh - Watt hours - is a better measure of battery capacity, as Mojo has two cells with a max total voltage of 8.4v. Your 5000mAh external battery is maybe only 18.5 Wh (assuming it is a single cell), and when you figure in inefficiencies in power delivery, you will need more than 18.5 Wh from a portable battery to fully recharge Mojo. 10000mAh single cell should be fine.
 
As other posters have said, 1A is min, 2A is fine but Mojo will not charge any faster. I have controlled the charge time for thermal reasons.
 
Rob

 
 
Quote:
JF here - Mojos multiple DSP cores and all other circuitry develop 1.7 Watts of heat when running this heat it dissipated from Mojos case through convection and heat radiating away. This can only happen when the Mojo cases temperature is a few degrees above the ambient temperature so it will feel warm in a high ambient environment. This is normal and totally safe as there are three separate and independent thermal sensing and protection circuits to look after Mojo and Mojos special battery.

 
JF here - ....  the battery is perfectly safe right up to 150 degrees Centigrade we had it made that way and its costs more than other batteries. The case of the mojo when it's charging has to shed about 1. 7 watts of heat it can only do that by radiating it away or convecting it away. If it's in a warm environment or it can't covect its heat away it's temperature will rise until there is a sufficient temperature differential to shed its heat. It's perfectly safe we have three indepentdant thermal temperature safety shut down method in mojo so please don't worry we know what we are doing. Remember a really hot cup of tea is usually only about 60 degrees C.

 
Mojo actually has three independant thermal cut outs a special high temperature battery and very sophisticated charging circuitry . Picking up on an earlier post Mojo actually does not dissipate a lot of heat when it's working. It's only about 1.7 watts and when it's charging it adds about another watt so its not much really. However the electronics and battery are thermally bonded to the aluminium case. The Mojo's case can only shed its heat through convection or by radiating it away. This can only work if there is a temperature differential between itself and its ambient surroundings if there is an insufficient gradient between them, the Mojos temperature will rise until there is a large enough difference to pass its heat to the air surrounding it.
If it is prevented from doing this perhaps by being insulated I some way it's temperature will rise until one of the three shut down trips operate. note the battery is safe to 150 degrees and the trips all operate up to a hundred degrees lower. Therefore it's perfectly safe. In fact if it's feeling mildly hot at first to your hand. Your hand alone will easily soon bring the unit down to a reasonable temperature.

 
 
   
.... The unit turns off after a while. Probably when it gets too warm. That was kind of a disappointment as I was going to use it as a DAC for my PC as well.

You get the most power loss when it is charged from red; and when its being charged at full blue then the power from charging is very small.
 
If you do need to charge & listen at the same time from red, and its in a hot room, then if you charge it with Mojo on its side so the top & bottom is in free air, it will not turn off. A head-fi poster mentioned this & it works well, as Mojo's power dissipation is almost doubled by doing it this way.
 
Rob

 
Quote:
  If you fully charge Mojo then use it in a desktop it will not switch off; the power dissipation that the charger uses in matching the current drawn by Mojo is negligible. You are only at risk when charging & using at red - & indeed as Mython says putting Mojo on its side will solve that issue too.
 
Just to give you some numbers - fully charged and matching Mojo's current draw the power dissipation is 107 mW for the charger circuit. That will increase running temperature by less than 1 deg C. But at flashing red it is 910 mW for the power dissipation in the charger.
 
Now I could fix this by using a switcher based charger rather than a linear one - but these inject too much RF noise onto the battery. This would impair sound quality, & Mojo's design goals was that plugging in the charger would have no significant change in SQ - which would not happen if I used a switcher based charger. I am not prepared to damage SQ as to me this is the most important aspect just for a tiny improvement in usability.
 
Rob
 

 
Quote:
Originally Posted by Rob Watts /img/forum/go_quote.gif
 
The noise is due to ripple voltage on the charger upsetting the inductors/capacitors within Mojo. If you use a clean quality PSU & a low resistance USB cable to the charger PSU the mechanical noise should be silent or insignificant. 
 
Rob

 
Quote:
Originally Posted by Rob Watts /img/forum/go_quote.gif
 
The problem is a noisy USB VBUS power line, & this makes the inductors in Mojo vibrate. Also, some USB cables have high resistance, & this makes the problem worse - so using a different cable can make the mechanical noise go away. The PSU itself can make it better or worse. Don't worry about it if you hear the noise, Mojo is not faulty & will continue to be reliable.
 
Rob

 
 
  is it normal to have the mojo make a hissing sound (ie the unit itself, audio output is fine) when plugged in to power? brand new unit but it makes this weird noise when its plugged in (both when off and on). the hissing dies down when i plug in the signal cable (unit still off), not sure if i should send my unit back.

It's normal - it is the charging regulator going into low power mode. Don't worry, there is nothing wrong.
 
Rob

 
Further testing on the charger and cable compatibility. Chosen Anker PowerPort 5 for good measured performance and multi-port charging with Anker PowerLine due to good construction with braid + foil shielding. All are available at a reasonable price.


Case 1: Virtually silent, only heard very minor hiss when ear is pretty much on the unit
Case 2: Only noticeable hiss when you put your ears near the unit
Case 3: Loud whine lasting the only the first few seconds, faint charging noise after that.
Case 4: Loud whine, and it goes on for a few seconds and off for a few (voltage drop causing charging circuit to shut down)




Combo 1: Sony or Samsung charger + Sony or Samsung USB cable + extension cord = Case 4

Combo 2: Sony or Samsung charger + Any cable = Case 3

Combo 3: Apple 1A charger + Sony or Samsung USB cable = Case 2

Combo 4: Anker charger + Sony or Samsung USB cable = Case 2, slightly quieter than combo 3

Combo 5: Sony or Samsung charger + long 6ft Anker cable (same cable length as Combo 1) = Case 3

Combo 6: Apple 1A charger + Anker cable/Chord Mojo's bundled cable = Case 1

Combo 7: Anker charger + Anker cable/Chord Mojo's bundled cable = Case 1, slightly quieter than combo 6 directly compared, with no pattern to the noise (always the same loudness)



Anker PowerPort is tested at it's worse case scenario(unloaded), the ripple and spike measurements are better when the charger is fully loaded with devices. The Anker charger have noticeable more steady noise pattern than the Apple charger even with the best cable connected, the Apple charger's noise ripples in loudness and the Anker one is very steady at the same amplitude.



In short:

If you already have a Apple charger handy, just getting a quality USB cable like the one I've tested will yield noticeable gain, especially if you are using longer cables.

If you don't have an Apple charger, getting an Anker charger with their PowerLine cables will yield the best possible result without going to go with a lab bench linear power supply. Having a multiport desktop charager will also allow you to run shorter cables.

Link where I bought them:
Anker PowerPort 5: http://www.amazon.co.uk/dp/B00VTI8K9K
Anker PowerLine: http://www.amazon.co.uk/dp/B014H3GKZ4


 
 
 

 
 
Interference is only noticeable when your cell is on 2G, & only if your IEM cable is within 3-6 inches of the phone (that seems to be where the interference comes from, the headphone jack)...

3G/4G/LTE etc do not have any effect that I can tell...

 
Quote:
Waaaaay back, early in the thread it was determined that 3G and LTE showed little to no EMI noise, but 2G/Edge cellular reception was very noisy. I heard no noise until I switched to 2G/Edge & it was brutally obvious. That's another factor besides cables acting as antennae.

 
  ok seeing everyone has earned a day off at The Pump it's down to me to bring you a 'Official' Chord-Electronics announcement dealing with RF interference with phones...
 
We at Chord Electronics suggest that people switch to flight mode when using Mojo Especially when it's strapped to a phone.
 
When using a phone as the source the RF noise level is dependant which frequency its on & how far the base station is away from the phone.
 
This is because the phone will ramp up its transmit power to make the connection.  This is almost like an EMP bomb going off right next to the Mojo and its two dangling RF receiving cables. We have taken all precautions, but there is little that can be done to overcome the massive amount of RF that is generated within millimetres of the Mojo, hope this helps in some way. 

 
  The problem lies in the cables that we attach both input & output & the variations thereof. These act as aerials feeding directly into Mojo. A phones level RFI in close proximity to mojo is very severe & therefore this issue is not easily solved without compromising Mojos performance this is because when a phone loses signal it ramps up the transmitt levels dramatically & these can be on any number of frequencies. Some cables are adequately screened & with those there is unlikely to be a problem, but with unscreened types there may be. That is why we recomend that for critical listening & in environments where a signal is likely to be lost that you switch to airplane mode.

 
  You can get various sizes of snap-to-fit ferrites at most electronics shops.
https://www.radioshack.com/products/radioshack-snap-choke-core?variant=5717355973
 
I'm not saying it will make a difference to the sound quality etc.

 
   
Basically, if you experience RF issues when Mojo is connected to a smartphone, then:
 
  1. ensure you are using a proper coaxially-shielded connector cable
  2. if available to you, try to operate your smartphone on 3G or LTE, & not on 2G/Edge
  3. try using the smartphone in 'Airplane' mode whenever possible
  4. Co-ax cables, & USB connector cables, can often be purchased with a ferrite choke manufactured integral to the lead, but if your cable does not include one, you may find it worthwhile obtaining a small ferrite choke to clip around the cable, locating it as close to the DAC end as possible
 
  1. Quote:
  We are hearing good things about Audio Quest jitter bug - I have heard it stops the mobile phone EMC problems that can happen with Mojo & certain headphones. I will be checking it out soon.
 
Rob
 

 
Where can I BUY Mojo in my Country?
 
 
Where can I BUY Mojo in the UK?
 
 
Where can I REGISTER my Mojo product?
 

 
Oct 14, 2015 at 8:56 AM Post #3 of 42,758

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Head-fi Mojo FAQ (started by 'Currawong') (NB: this should be considered supplementary to the comprehensive information, below)​
 
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  Here are my slides from the Shard presentation:
 

 

 

 

 

 

 

 

 
 

 
 
 
Quote:
  Its a brain issue, and is (mostly) down to two technical problems - one being noise floor modulation, one being timing uncertainty. With timing uncertainty, when the sampled digital data is converted back to a continuous signal, the DAC creates timing errors. These timing errors then interfere with the brains ability to actual perceive the starting and stopping of notes - and when the brain can't easily recognise something, it has to work harder to make sense of what is going on. Its a bit like one being in a party trying to understand somebody speaking with a lot of noise - your brain has to work harder to understand the voice, and its tiring. The noise floor modulation problem, means that the brain has greater difficulty separating sounds out into individual entities. What people forget, as we take hearing for granted, is that the brain is processing the data from the ears, and separating things out into individual entities, and also putting a placement tag onto that entity. Noise floor modulation makes it more difficult for the brain to separate things out into individual entities, so the brain has to work harder to make sense of the music. And when it has to work harder, you get listening fatigue.
 
Now the timing issue is a unique problem with digital audio, and noise floor modulation is about ten times a larger problem than with amplifiers, so you can see why listening fatigue is a particular problem with digital.
 
Rob
 
Quote:
 
Is it my brain. Or is it that Mojo just gets better and better the longer I use it? This little gadget is really stunning.

I had same experience with Hugo. It just seemed to get better and better, and took 9 months before the feeling of improvements stopped. Funny thing was it was not break in as new Hugo sounded the same. My assumption was my brain breaking in to the way that Hugo recreated transients which was quite different to any other dac before.
I expect Mojo to be the same.
Rob
 
Quote:
  Electrolytic capacitors take time to break in - leakage current takes 3 months to minimize and so does ESR (equivalent series resistance). If you use them in the audio path (I do not) then bass distortion gets lower with time. It is possible to reduce break-in time, and I do this.
 
With Hugo I kept on getting the feeling that SQ was getting better and better - even nine months on - but when given brand new product from Chord, once warmed up, they sounded the same. So it was not the hardware, and either I was deluding myself, or my brain was un-learning digital music. Now Mojo/Hugo/Dave do things in the time domain that no other DAC's do, so its easier for the brain to make sense of the music as timing of transients has much less uncertainty. Certainly the brain does get used to a particular sound, and creates processing short cuts that allows better understanding of the sound, so its not a great leap to state the possibility that our brain's unlearn digital sound as after all, we are surrounded by it.
 
I can say that since Hugo I can no longer tolerate listening to music using conventional DAC's.
 
Rob  

 
Quote:
  The brain breaking in problem I think is actually about us dealing with conventional digital audio - we listen to digital music all the time - TV, Hi-Fi and portable gear and actually listen to un-sampled music probably less than 1% of the time - so our poor brains is saturated by having to deal with the timing problems of sampled digital music - and I guess it has created coping strategies to deal with uncertainty in the timing of transients. Then along comes something different, with the timing uncertainty removed - so the brain has to unlearn the coping mechanisms.
 
Now I am of course only guessing here, but it was very odd when I first heard Hugo - and that took 9 months for me to get used to the sound - I had this constant feeling that SQ was getting better - it was not Hugo, as new units sounded the same as my old unit.
 
Having said all that, the first ten seconds of listening to Hugo I knew immediately that something very strange was going on, as the sound was very different to what I was used to, so people should hear the difference that Mojo makes very quickly. But I have the benefit of being an experienced and sensitive listener.
 
The really curious thing about all this is that the actual timing differences in terms of error is very small - the ear/brain is a remarkably sensitive system, and science has little understanding about things we take for granted, such as out perception of sounds. How does the brain separate sounds out into discrete entities and put an extremely accurate placement tag on it? There is some amazing processing going on for which we have no understanding.
 
Rob  

 

Also see John Franks' remarks: https://youtu.be/DTWcKLI0g7c?t=41m7s
 
 
 
  I got my Mojo yesterday, and have been comparing to my iDSD micro. I hear what people refer as "musical" and "emotional," but I want to understand why. The iDSD sound more flat, more monotone almost compare to the Mojo. Is the Mojo more true to the original sound? I'm not sure, but it sure sounds more fun. It's almost like describing a picture, the Mojo has more contrast and vibrancy, where as the iDSD has a more flatter, but possibly more true color.
 
Then there's the matter of warmth. The iDSD is known for its warmth, yet that warmth has a bit of way veiling the sound. Mojo also has a hint of warmth, but that warmth is applied without extra veil. I'm in no way dismissing the iDSD, as it is absolutely superb, and perhaps more neutral. But the Mojo just sounds more exciting than the iDSD. I'm comparing using both my HD800 and the Mainline as amp, as well as my SE846 directly plugged into the Mojo/iDSD unit.
 
One thing for sure though, the Mojo is much more compact and portable than the iDSD. Too bad I'm not using them as portables and strictly as desktop DAC's. The iDSD is more desktop friendly and allows easier charging while being used. I haven't tried both charging the Mojo while it's being used, but I read that it uses battery faster than being charged, being in the desktop mode. Since it has more settings, the iDSD is also more compatible with a wider range of headgear, be it IEM or high impedance full sized headphones. I just love the build on the Mojo though, it's truly a work of art.

 
 
 
Relating sound quality to technical performance is very complex, and I will try to explain, but I could talk for days about it and completely confuse everybody. But here is a quick answer to your questions.
 
 
   I hear what people refer as "musical" and "emotional," but I want to understand why.

 
Musicality and emotional is complex, but in a nutshell its about removing distortions that interfere with the brains ability to understand the music. Conventional DAC's have a number of distortions that make it much harder for the brain to perceive the sound. Now we underestimate what the brain does with hearing, and simply make the assumption that the ears convert sounds into nerve impulses, and that's that job done, the brain simply access's the nerve signals. But that's not what happens - audible reality is an illusion created by the brain, and a considerable amount of brain processing is employed to create that illusion. So for example, you listen to a guitar and a singer for example. The data the ears feed the brain is a jumbled up mess of information, and the brain separates this mess of data into two distinct entities - the guitarist and the singer, and you perceive this as two separate entities. Not only that, but the brain very cleverly calculates where in space those entities are, and it does this from subtle timing, amplitude and resonance cues from both ears. But this requires considerable calculation. Moreover, small and subtle distortions (by saying distortion I mean anything that changes the original signal in any non linear way) interferes with the brains ability to separate sounds out into distinct entities, and interferes with the brains ability to place entities in space. This has two consequences for being able to enjoy music - firstly the brain is struggling to process the data, so has to work harder - which means you get listening fatigue, and so you can't enjoy the music. Secondly, being able to enjoy the music means being able to perceive what is going on - and there are many distortions that disable the brains ability to perceive the music. This is where it gets complex, as there are a myriad of different distortions that upset the brains processing. That's why Mojo has the WTA processing, why it filters and over-samples at 2048 times, why its got noise shapers that are a thousand times more resolving than conventional noise shapers - I could go on.
 
 
 
  The iDSD sound more flat, more monotone almost compare to the Mojo.

 
The perception of depth information is down to very small amplitude differences of small signals. Now the brain calculates depth from a number of different cues, but most of it comes from the reverberant sound from the acoustic the recording was made in (or depth is added by adding artificial reverb). Now reverb is very small signals, and the amplitude accuracy of these small signals is crucial for the brain's ability to calculate depth. Now there is something very strange about depth perception - and that is the brain needs these small signals to have perfect amplitude linearity. If a small signal is slightly larger or slightly smaller than it should be, then the brain gets confused and can't calculate the depth properly, and things then sound flat. But the amazing thing is, there appears no limit to how accurate these small signals need to be in order for the brain to not truncate or flatten depth. In order to accurately reproduce depth you need extreme small signal linearity. You can't do this with R2R DAC's, as the resistors can't be matched. With DSD or delta sigma (Mojo is delta sigma too) the problem is now how well the noise shaper functions. As a signal gets closer to the noise shaper noise floor, the levels get smaller, as a signal that is smaller than the resolution limit of the noise shaper is truncated. To overcome this you need to have very high resolution outputs, with a noise shaper that has very high resolution - in Mojo's case, the noise shaper has a thousand times more resolving power than conventional high end noise shapers, and ten thousand times more resolution than DSD 64. But there is another source of error that can upset sound-stage depth and this is digital noise adding to the analogue signal. This applies to all DAC's, and is a big problem with chip DAC's, as there always exists a path from the digital noisy part to the analogue part, and this noise corruption will degrade the small signal non-linearity. But with Mojo the actual analogue parts are discrete, so its possible to eliminate digital noise from corrupting the signal. There is another mechanism for depth to be truncated, and this is with metal to metal interfaces. When you have a soldered joint, or any metal to metal interface, oxides and impurities concentrate at the interface. This oxide barrier is non-linear in that the resistance to small signals is larger than with big signals - so again we have small signals being attenuated. To reduce this problem you can only do this by reducing the number of passive components in the signal path. Conventional DAC's (delta sigma and R2R) have very complex analogue components, due to the need to convert from differential to single ended and to filter the high amounts of RF that comes out of a conventional DAC. With pulse array (my DAC technology within Mojo) this is not an issue as I can get single ended to work, and it runs at 104MHz, so little analogue filtering is required.
 
 
 
  The iDSD sound more flat, more monotone almost compare to the Mojo.

 
I think here you are referring to timbre - the tonal colour of the instrument. Now timbre is an issue with timing reconstruction, as the brain uses transient information to infer the timbre of an instrument. Now conventional digital has uncertainty in the timing of transients (does a signal cross through zero just after a sample, or in the middle or close to the end of a sample?) and the only way of recovering the timing information perfectly is to use an infinite amount of processing on the interpolation filter. With the use of the WTA filter, which has been optimised to recover timing, and 500 times more processing than conventional DAC's, I can reduce the timing uncertainty - which results in much better timbre variation, so things don't sound monotone. There is another aspect in that noise floor modulation also affects timbre reproduction, but this is answered in your next question.
 

 
  Then there's the matter of warmth. The iDSD is known for its warmth, yet that warmth has a bit of way veiling the sound. Mojo also has a hint of warmth, but that warmth is applied without extra veil. I'm in no way dismissing the iDSD, as it is absolutely superb, and perhaps more neutral.

 
Warmth or smoothness can be artificially created - for example with a dollop of 2nd harmonic. Mojo has very low levels of distortion, so its warmth is not down to doing this or other things. The key to true refinement with DAC's is noise floor modulation. This is where the noise pumps up and down with the signal, and all other non Chord DAC's have large amounts of noise floor modulation. Now noise floor modulation is a scary issue with DAC's, and there are countless ways that a DAC can suffer. Mojo, on the other hand has zero measurable noise floor modulation - the noise floor is at -170dB and it maintains this whether its output is 2.5v or zero, the noise is completely static. Now the issue of neutrality is a very complex thing, as increasing transparency will make it brighter and sharper, and increasing refinement will make it smoother and darker, and its possible to use distortion to create the impression of warmth or brightness. To be honest, I (or anybody else for that matter) do not know what the tonal balance of a perfect (and hence neutral) DAC is. And neutral cam mean different things to different people and with different gear!
 
Mojo's musical performance is down to lots of technical things - way too complex to talk in detail with - but there are solid reasons why you hear what you hear, and why other DAC's can't do this.
 
Rob


 
 
Quote:
  There has been some recent discussion about digital filters, in particular closed form mathematics. There is a lot of confusion about what is actually happening, and this is not surprising - filter design is complex, and people talk about things that they have little real understanding.
 
Indeed, the more time and work I spend in audio, the more I realise how much more there is to know - we are all scratching at the surface, so some humility is needed. "You know nothing Jon Snow" is my favourite quote from Game of Thrones, and I often bear it in mind when thinking about audio, and how to relate something I hear with theory.
 
Now there are two things that are talked about closed form filter design - one being that the the filter coefficients (these are fixed at the design of the filter) uses a closed form algorithm which just means that it is a formula to calculate the numbers. The second issue is that the initial filter samples are preserved.
 
Now most FIR filter algorithms are closed form. The exception, as pointed out by a poster earlier is the Parks–McClellan which uses the Remez algorithm to iteratively calculate the optimal solution for the coefficient calculation. It is not a closed form calculation, as it cleverly runs backwards and forwards until it converges onto the desired result. Now is a closed form a good or a bad idea? Frankly, it does not matter how the coefficients are calculated, its what those coefficients are, and what they sound like that is important. Now I don't like the Parks-McClellan algorithm, as it does not maximise rejection at the points where there is the most out of band energy which is at FS multiples. And its not very good at recovering timing information for the intermediate samples you are trying to create. But this is not closed form or iterative process that is important here. Now the WTA algorithm is closed form, you can calculate the ideal coefficients to as much accuracy as you like with one fixed equation. But whether it is closed form or not is just unimportant.
 
The second issue is exactly maintaining the original samples. Now the vast majority of FIR filters for audio are known as half band filters, and to create a 8 times oversampled filter you use a cascade of 3 half band filters. These are guaranteed by design to give the original data, and they are used because they are computationally efficient, as half the calculations are zero - you simply return the original sample, no maths. Most are designed with Parks-McClellan, so the issue of closed form has actually nothing to do with retaining the original sample data.
 
So maintaining the original sample data is a red-herring as regards closed form. But is keeping the original data actually a good idea? It sounds like a great idea, why mess with the actual data?
 
When I was developing the WTA algorithm in the late 1990's I hit a stumbling block. I had designed a very long tap length half band filter - so it was 2048 taps, half being zero, so it returned the original sample perfectly. It sounded very much better than the filters I had before, but I knew that timing recovery and transient accuracy was a problem. I could see also that aliasing issues from the half band filter would degrade transient accuracy, so I needed to remove these measurable aliasing problems. But that would mean the original data would get changed, and I did not like that.
 
One trap that designers and audiophiles fall into is to think doing XYZ is wrong and that it must sound better because of this particular idea. That is a very easy trap to fall into - or even think some idea must sound better, then listening to it, then convincing yourself that this soft muddled sound is actually better (or this bright hard sound is more transparency and at last I can hear how bad recordings actually are). In other words your thinking is convincing yourself that something is better (of course your lizard brain is not fooled and you end up listening to less music and enjoying it less). I too was stuck in the trap that the best thing to do was to keep the original data. But at the end of the day, you got to try it, do careful listening tests, and run by the evidence, not what you think may sound good or con yourself into thinking something is better. So eventually I tried eliminating the reconstruction aliasing, and boy did this make a big improvement - even though the samples were not being preserved - bass was much deeper, sound-stage much more accurate, and the flow and timing much more natural. 
 
So some humility is called for, nobody has a perfect understanding of anything, and thinking something must sound better is extremely dangerous. Do the work, listen carefully and neutrally, and base everything on the evidence, not on attractive ideas.
 
Rob

 
  PC's are very restricted in what they can do for real time signals. You simply can't replicate the processing that Dave does in a PC - simply because PC processors are sequential serial devices with a very limited number of cores. When you are doing a doing a FIR filter (a tap) you need to read from memory the audio data; read from memory the coefficient data; multiply the numbers together;then read the accumulated data and add that to the previous multiplication; then save the result. Lots of things to do in sequence. With an FPGA you can do all of these things in parallel at once, so a single FIR tap can be accomplished within a single clock cycle (obviously pipelined) - you are not forced to do things in sequence.
 
With Dave I have 166 dsp cores running, plus FPGA fabric to do a considerable amount of further processing. You simply can't do that in a PC. To give you another example - converting DSD into DoP. You need a quad core processor to do this manipulation in real time - otherwise you get drop-outs - but in a FPGA I could do this simple operation thousands of times over, and at much faster rates than DSD256.
 
What some people do not understand is how capable FPGA's are and how widespread they are used - the backbone to the internet? FPGA's. Search engines? FPGA's. Why? because an FPGA is fantastic at doing fixed real time processing - it takes small die area, and can do complex operations with very low power. Mojo for example has 44 dsp cores, uses sophisticated filtering to 104 MHz, and noise shapes at this rate - but does all this whilst consuming only 0.45 W. There is no way any PC consuming huge amounts of power can do this.
 
Intel last year acquired Altera (an FPGA company) for $16.7 billion because they understand that the future of processing is with FPGA's
 
A second issue is not what you can do but how you can do it - it is not just about raw power, but how the filter algorithm is designed. I have put many thousands of hours and over twenty years improving and understanding how to make a transparent interpolation filter; and I am still learning things today.
 
And a third point is that a DAC is not simply a data processing machine but it has got crucial analogue parts too. If I dropped the WTA requirement, I would still need the same FPGA in order to do the noise shaping and other functions.
 
Rob

 
 
 
@robwatts @mojo ideas

How about an impedance module that allowed us to adjust output impedance until we perfectly matched mojo to our ciems/headphones?

The technically perfect impedance is zero, and that's why I worked so hard to get it as low as 0.075 ohms with Mojo.
 
The reasons going for as close as zero are:
 
1. Frequency response. The impedance of the headphone varies with frequency, and so by having a high output impedance will cause frequency response variations. Zero impedance eliminates this problem.
 
2. Distortion. The impedance of a headphone varies with level, and having a higher output impedance will increase the total distortion - given that Mojo distortion is so low, this is actually quite a significant an effect. Again, zero impedance eliminates this problem.
 
3. Damping factor - probably the most important reason. A drive unit is a resonant system - that is a mass on a spring - that is damped mechanically and electrically. Electrical damping is due to the headphone creating a current due to the motion of the driver in the magnetic field - and how well this is controlled depends on the electrical impedance the driver sees - in our case, the cable impedance and Mojo's impedance. Again, zero impedance gives the best damping, with an infinite damping factor.
 
I did some listening tests many years ago with loudspeakers and damping factor and found that it made a massive difference to the sound. Damping of 10 gave a very soft, big fat bass - but everything sounding one note in the bass - simply because the loudspeaker was doing its own thing at the resonant frequency. Going from 10 to 100 gave a tighter bass, with much better pitch reproduction - you could follow the bass line much more easily. Above 100 to 1000 it sounded tighter - no big change in pitch (being able to follow the bass tune) but the perceived tempo of the music became faster as transients are much better controlled. Going above 1000 gave a small improvement in how tight it sounded.
 
Rob

 
 
 
 
What does *laid back* mean in sound? I never understood. Is mojo considered laid back
?

 
Laid back or relaxed tends to suggest less dynamic punch and a less forward sound. Less in your face is another way to put it. This can also mean less fatiguing.

 
OK I appreciate it, it makes sense, I like my sound a little punchy and in your face, but also not fatiguing haha, mojo definitely *warmed* up my sound and I enjoy that, because it still stays very clear and detailed

I often think about this issue as yin-yang (dark-bright), and a good product has this in balance - but what the correct balance is does depend somewhat on taste!
 
So yin - dark - is in technical terms, happens with zero noise floor modulation. Conventional DAC's have enormous levels of noise floor modulation. This means noise (bright hiss) pumps up and down with the music signal, and the brain can't separate a dark sounding instrument from the noise floor modulation - so smooth sounding instruments become bright. With Chord DAC's, including Mojo, there is no measurable noise floor modulation, so it innately sounds smooth and warm.
 
But its possible to artificially give the appearance of more yin by contouring the sound. For example, add a lot of second harmonic distortion, and it sounds thicker and darker - but its an illusion, as everything sounds soft. You can also add LF errors too, to give the impression of more weight to the sound - adding electrolytic caps, or letting the ref circuitry amplitude modulate the output from the signal envelope. Indeed, a lot of designers rely on this, as they do not have the abilities (stuck with using chip DAC's) to solve noise floor modulation, so have to use tricks to balance the sound.
 
On the yang side, natural brightness comes from two sides. First is transparency, and this resolves into detail resolution, and this is about how accurate the DAC/amp can resolve very small signals accurately. With my work on the reference DAC Dave, I discovered that there is no limit to how accurate the small signal needs to be - the smallest possible amplitude error is very audible, particularly in terms of sound-stage depth. Transparency is a complex issue, but comes down to two main issues - simplicity of the analogue section (each component degrades small signal linearity) and the performance of the noise shaper (before anybody says ladder DAC's these are awful for small signal linearity). Now Mojo has an extremely simple output stage - only one active stage and two resistors and two capacitors in the direct signal path, and this is done for transparency. On the noise shaper, it has 1000 times more resolution than conventional noise shapers, as the noise shaper runs at 104 MHz, not the usual 6 MHz of the best chip DAC's.
 
The second part of yang is timing. Now digital audio is sampled data, but the original signal in the ADC is a continuous signal, and the job of the DAC is to convert the sampled signal into a continuous analogue signal with the timing of the original signal in the ADC perfectly preserved. Now I talk a lot about reconstituting timing, and have had requests to show the problem. So here is a simple illustration of the problem:
 

 

 

 
Now this is a bit of a simplification - the burst signal is not bandwidth limited, but it serves to illustrate the problem of timing inaccuracies. Now how do these timing errors sound like? When the brain comes across timing errors, it can't deal with it - it can't make sense of the music. And when the brain can't process the signal, you then can't hear the transients. It is a bit like putting a picture out of focus, blurring the edges. What this does audibly is to make transients sound soft, and when one improves timing accuracies then the brain can perceive the starting and stopping of notes accurately - so things sound sharp and fast - more yang. Now what is curious about timing errors, is that there again is almost no limit to how small they need to be - before Dave, I used to think in terms of uS errors, now its definitely nS as being important - extremely small timing errors have a noticeable subjective musical impact. 
 
 
Also it is very possible to use distortions to give impressions of good sound - use slew related noise floor modulation and you get the impression of good timing resolution - but its entirely false. The problem with using distortions like this, although it can sound superficially impressive - is that everything always sound the same. But the major problem with this approach is simply listening fatigue - I can listen to Mojo for 10 hours and still want more. It also illustrates the design nightmare of listening tests - is the sound quality "improvement" real or just more distortion or aberration? You have to be extremely careful on how one assesses sound quality.  
 
So to conclude - Mojo can sound both rich & dark (immeasurable noise floor modulation) & very fast & dynamic (much lower timing errors) all at the same time. That's why we get so many different reactions to the sound of Mojo - some saying its rich & smooth, some saying its fast and dynamic - & the truth is both observations are correct.
 
Rob

 
 
 
Quote:
 
  Digital transmission is based on SPDIF standard which transmits data and clock information as an encoded signal usually using PCM, that information is decoded on the Mojo into data and clock signal so it's important that the encoded information be jittered free and not degraded over short distance.
 
The USB transmission on the other end is a device to device transmission mechanism using an encoding scheme and handshaking mechanism, it is usually stream based so more tolerant to poorer wire as frames are transmitted and decoded from the source to the target device. The target device will reconstruct the data and clock signal from the frame and then feed it to the DAC to be analog reconstructed and eventually band pass filtered to remove any residual high and low frequency signals out of the audio band.I still think you need to keep the USB cable short but it is more tolerant of longer lengths up to a limit.
 
To make a story short, the short USB cable is fine but an analog cable used as a digital one is just a bad idea. Again, that's just my opinion.


Just to clarify:
 
1. SPDIF decoding is all digital within the FPGA. The FPGA uses a digital phase lock loop (DPLL) and a tiny buffer. This re-clocks the data and eliminates the incoming jitter from the source. This system took 6 years to perfect, and means that the sound quality defects from source jitter is eliminated. How do I know that? Measurements - 2 uS of jitter has no affect whatsoever on measurements (and I can resolve noise floor at -180dB with my APX555) and sound quality tests against RAM buffer systems revealed no significant difference. You can (almost) use a piece of damp string and the source jitter will be eliminated.
 
2. USB is isochronous asynchronous. This means that the FPGA supplies the timing to the source, and incoming USB data is re clocked from the low jitter master clock. So again source jitter is eliminated.
 
So does this mean that any digital cable will do?
 
Sadly no. Mojo is a DAC, that means its an analogue component, and all analogue components are sensitive to RF noise and signal correlated in-band noise, so the RF character of the electrical cables can have an influence. What happens is random RF noise gets into the analogue electronics, creating intermodulation distortion with the wanted audio signal. The result of this is noise floor modulation. Now the brain is incredibly sensitive to noise floor modulation, and perceives this has a hardness to the sound - easily confused as better detail resolution as it sounds brighter. Reduce RF noise, and it will sound darker and smoother. The second source is distorted in band noise, and this mixes with the wanted signal (crosstalk source) and subtly alters the levels of small signals - this in turn degrades the perception of sound stage depth. This is another source of error for which the brain is astonishingly sensitive too. The distorted in band noise comes from the DAP, phone or PC internal electronics processing the digital data, with the maximum noise coming as the signal crosses through zero - all digital data going from all zeroes to all ones. Fortunately mobile electronics are power frugal and create less RF and signal correlated noise than PC's. Note that optical connection does not have any of these problems, and is my preferred connection. 
 
Does this mean that high end cables are better? Sadly not necessarily. What one needs is good RF characteristics, and some expensive cables are RF poor. Also note that if it sounds brighter its worse, as noise floor modulation is spicing up the sound (its the MSG of sound). So be careful when listening and if its brighter its superficially more impressive but in the long term musically worse. At the end of the day, its musicality only that counts, not how impressive it sounds.         
 
Rob
  The reasons why sources and digital interconnects sound different are well understood - see some of my posts. In a nutshell it is not jitter (all my DACs are completely immune to source jitter) but down to RF noise and distorted currents from the source flowing into the DAC's ground plane. The RF noise inter-modulates with the analogue electronics, creating random noise as a by product, which creates noise floor modulation, and that makes it sound brighter or harder. The correlated or distorted currents very subtly add or subtract to small signals, thus changing the fundamental linearity, which in turn mucks up depth perception.
 
But I also agree in that lots of people hear changes that are not there - I for one have never heard any difference with optical cables (assuming all are bit perfect) with my DAC's, but lots of folks claim big differences. Placebo, or listening with your wallet, plays a part here. Then there are cases of people preferring more distortion... Listening tests must be done in a very controlled and careful fashion, particularly if you are trying to design and develop things.
 
Rob

 
 
At the risk of being flamed, I don't see how the composition of the USB cable wire can add warmth to digital data from whatever device is being used as a transport

I understand those concerns too - after all the data is the same. But there are solid scientific reasons why they can make a difference.

 
In the 1980's, people started talking about mains cables making a difference to the sound quality - and I didn't believe it either - particularly as my pre-amp had 300 dB of PSU rejection in the power supply. But I did a listening test, and yes I could hear a difference. Frankly I still could not believe the evidence of my own ears, so did a blind listening test with my girl friend. She reported exactly the same observation - mains cables did make a difference to SQ.
 
To cut a long story short, I proved the problem was down to RF noise. RF noise inter-modulates with the wanted audio signal within the analogue electronics, and if the RF noise is random, then the distortion is random too and you get a increase in noise floor with signal. This increase in noise floor is noise floor modulation, and the brain is very sensitive to it; you can perceive tiny amounts of noise floor modulation as a brightening or hardening of the sound. By tiny I mean the noise floor modulation needs to be well below -200 dB, so the brain is very sensitive to it. With the right test equipment, you (APX5555 is only test equipment that has no innate noise floor modulation) can easily measure the effect.
 

The RF characteristics of the cable can change the RF noise that gets injected into Mojo's ground plane, and this is the mechanism for changes in smoothness. You may say why can't you make it insensitive to it; well I go to silly lengths to RF filter and decouple, and use dual solid ground planes on the PCB, but you can't remove the problem. For Dave, Hugo TT and 2 Qute I have galvanic isolation, and this eliminates the problem (along with other SQ problems such as sound-stage depth). But I can't do this with portable devices, as it draws power from the 'phone. That said it's less of an issue with portable electronics as they are less power hungry and create less noise.

 

So what are the best USB cables? Firstly, be careful. A lot of audiophile USB cables actually increase RF noise and make it sound brighter, and superficially impressive - but this is just distortion brightening things up. Go for USB cables that have ferrites in the cable is a good idea - it may also solve any RF issues from the mobile that you may have too.

 

Rob

 
 
Quote:
 
Clock jitter -- What is clock jitter? The reason I ask is that in considering different DAPs to use as transports to output a digital signal to the Mojo, I've seen some varying specs for clock jitter on different DAPs, as follows:

AK100 90 ps (pico seconds)
AK120 50 "
AK240 50 "

Question: Does clock jitter degrade the digital signal before it's send out from the DAP? Or are they referring to clock jitter of the internal dac, in which case clock jitter doesn't matter since the signal never reaches the dac (it's been output beforehand)?

If clock jitter degrades the signal before it's sent out, then it appears that the AK100 is not as good a transport as the other two. But would the difference be discernible?

Thx

 
Clock jitter is timing uncertainty (or inaccuracy) on the main clock that is feeding the digital outputs. Its often expressed as cycle to cycle jitter as an RMS figure, but can be total jitter which includes low frequency jitter too. Total jitter is the most important specification. If you want here is a good definition:
 
https://en.wikipedia.org/wiki/Jitter
 
As you can see, the jitter subject can get complicated and its often abused by marketing...
 
But with all of my DAC's you do not need to worry at all about source jitter, so all of the above AK numbers are fine. So long as its below 2uS (that is 2,000,000 pS) you are OK, and nobody has jitter that bad!
 
Rob
 
Quote:
  .... Mojo has the same USB as Dave - but with Dave it is galvanically isolated, so RF noise and signal correlated currents from the source can't upset Dave at all. I can't do this with Mojo as it draws too much power from the device connected to the USB - but the upside is that mobile sources create much less noise as they are power efficient and there is no ground loops either due to battery operation.
 
So Mojo like Dave has solved the jitter problem via USB, as the timing for the data comes from Mojo not the source.
 
Rob

 
Quote:
  The reasons why sources and digital interconnects sound different are well understood - see some of my posts. In a nutshell it is not jitter (all my DACs are completely immune to source jitter) but down to RF noise and distorted currents from the source flowing into the DAC's ground plane. The RF noise inter-modulates with the analogue electronics, creating random noise as a by product, which creates noise floor modulation, and that makes it sound brighter or harder. The correlated or distorted currents very subtly add or subtract to small signals, thus changing the fundamental linearity, which in turn mucks up depth perception.
 
But I also agree in that lots of people hear changes that are not there - I for one have never heard any difference with optical cables (assuming all are bit perfect) with my DAC's, but lots of folks claim big differences. Placebo, or listening with your wallet, plays a part here. Then there are cases of people preferring more distortion... Listening tests must be done in a very controlled and careful fashion, particularly if you are trying to design and develop things.
 
Rob

 
Quote:
  There are two problems that USB has against toslink - and one benefit. The benefit is that timing comes from Mojo - but with toslink the incoming data has to be re-timed via the digital phase lock loop (DPLL) and this is not quite as good - but you will only hear the difference via a careful AB test, so it's in practice insignificant.
 
The downside with USB is the common ground connection. This will mean RF noise will get into Mojo, making noise floor modulation worse. Now I go to very careful lengths to remove this problem by using lots of RF filtering, and double ground planes on the PCB, but even minute amounts of RF is significant. The other problem is down to the way that digital code works - which is in twos complement. So zero is in 24 bits binary is 0000_0000_0000_0000_0000_0000. If the signal goes slightly positive then we get just one bit changing to: 0000_0000_0000_0000_0000_0001. But if it goes 1 bit negative all the bits change to:  1111_1111_1111_1111_1111_1111. Now the problem with this is that when a bit changes, more power is needed, and this injects current into the ground of the PC - and the ground will get noisier. Unfortunately the noise is worst for small signals. Now the problem with this is that it then couples through to Mojo's ground plane, and the distorted signal currents will add or subtract to small signals - thus changing the small signal linearity. This in turn degrades the ability of the brain to re-create depth information, and so we hear it in terms of depth being flattened. What is really weird about depth perception is that there seems to be no limit to how accurate it needs to be, so the smallest error is significant.
 
So with toslink we do not get these problems as there is no common ground - so no RF noise, no distorted signals on the ground, and it will sound smoother with better depth against a noisy PC. But the problem can be almost eliminated by using a power efficient USB source that is battery powered - such as a mobile phone. But with noisy PC's the only way of solving it is to use galvanic isolation on the USB - but this draws power from the source, and we can't do that with mobile devices. All of Chord's desktop DAC's have galvanic isolation on the USB, and then you can't hear whether its a noisy PC or a mobile phone. In this case, USB sounds slightly better than optical, because we have the (tiny) timing benefits of USB.
 
I hope that explains - its a complex subject.
 
Rob

 
 

 

 
 
  .... the volume control is in the central WTA filter core, and has an internal accuracy of 51 bits. But it then gets passed to the cross-feed dsp, then on to the 3 stage interpolation filters to take it to 2048FS, then into the OP noise shapers. So the 51 bits has to be truncated. But since the signal is at 16FS, the truncation is done via noise shaping and dithering. This means that the signal is not lost, but perfectly preserved, as this process adds zero distortion - just a fixed noise at -180dB. This has been verified with Verilog simulation.  
 
Rob

 
 
  Just to make it 100% clear - the USB input will measure absolutely identically to the coax or optical inputs if the USB data is bit perfect.
 
I have set up my APX555 so that it uses the USB via ASIO drivers, and I get exactly the same measurements on all inputs - 125 dB DR, THD and noise of 0.00017% 3v 1k 300 ohms. I have done careful jitter analysis, FFT analysis down to Mojo's -175dB noise floor, and can measure no difference whatsoever on all inputs (with the APX always grounded on the coax).
 
If somebody does measure a difference its down to mangled data on the USB interface (or perhaps poor measuring equipment - Mojo is way better than most test equipment). Mojo can't convert 16 bit data back to 24 bit....
 
Rob 

  .... Mojo has the same USB as Dave - but with Dave it is galvanically isolated, so RF noise and signal correlated currents from the source can't upset Dave at all. I can't do this with Mojo as it draws too much power from the device connected to the USB - but the upside is that mobile sources create much less noise as they are power efficient and there is no ground loops either due to battery operation.
 
So Mojo like Dave has solved the jitter problem via USB, as the timing for the data comes from Mojo not the source.
 
Rob

 
Also relevant:
 
 
 
 
Optical will support DSD with DoP on the optical - but only DSD64. Also, if you use a plastic fibre, only use very short lengths - for longer lengths you need a quality glass fibre. Running optical at 192 kHz is close to the edge for some optical transmitters and cables.

Rob

On a related note, for a given bitrate, is there any reason to expect optical cables of different build qualities to differ in SQ? Assuming they're capable of supporting the bitrate without noticeable artifacts, such as, intermittent "pops". Also, how big a role does the DAC implementation play?

Thank you for your opinion.

My listening test revealed no SQ change at all - so long as the data is arriving is still bit perfect. But with optical when it fails, it is fairly easy to spot bit failures. Of course, YMMV, and I guess if it's about to fail, you would hear an improvement with the odd bit error improvement. But I have not been able to hear a difference in my setup using plastic or glass.
 
Why would that be? Optical actually does not have bad jitter performance; but what it does do is have uneven rise and fall times. But my digital SPDIF receiver actually measures uneven rise and fall times, then uses that measurement to compensate to extract the data correctly. And as regards jitter - the DPLL completely removes jitter from the incoming stream - I can add 2uS worth of jitter, and see absolutely nothing coming out from Mojo with measurements - and Mojo's FFT noise floor is at -170 dB. So optical typically has 2nS of jitter, so that is a thousand times lower than a level that is still not detectable, even when I can resolve -170 dB.... So there is no technical explanation why it would make a difference. So if you do hear a difference, it is either because it is not bit perfect and has data errors (almost impossible with 44.1 though), or your suffering from a placebo (it looks nicer/costs more/must be better). 
 
Getting to your last point - the DAC has a big impact on this; most DAC's are very sensitive to jitter as they use analogue PLL techniques and they can't eliminate the jitter problems. So optical cables may have a SQ difference with other DAC's.
 
Rob

 
 

Is there a reason that different transports sound clearly different though, and more so on the mojo than on the Hugo? Not comparing between different inputs. The AK380 sounds clearly better to my ears than the AK100 does as as a transport. Among coaxial players the soundaware Esther m1pro sounds much better than most of the competition, and when the digital coaxial mode is activated and the amp and dac section are switched off it sounds even better.

Any possible thoughts on the reason for this?
smily_headphones1.gif

For electrical inputs - transports can make an audible difference for couple of reasons. RF noise from the source injected into the DAC ground plane will cause increased noise floor modulation; and the ear/brain is sensitive to minute levels of noise flloor modulation, so this is important - it will make it sound brighter with more noise floor modulation, and warmer and smoother with less. Additionally, depth and detail resolution is can be degraded by very tiny signal related but distorted currents; and this will subtly change small signal fundamental linearity (this is where small signals amplitude varies with signal level) and the ear/brain is incredibly sensitive to this; the smallest possible change in small signal resolution or accuracy will degrade the perception of depth. So very tiny distorted signal related currents will damage depth perception.
 
This is why optical is good; it does not suffer from any of these problems, as it is perfectly galvanically isolated.
 
And it explains why ASIO sounds better than WASAPI as less processor activity so less noise and hence better sound - so anything that you do with an electrical connection that reduces RF noise such as less processor activity or less power consumption may have a small benefit.
 
I have not noticed that Mojo is more sensitive than Hugo; if anything I would guess at maybe the other way around!
 
Rob

 
 
  Converting the original file into DSD or up-sampling is a very bad idea. The rule of thumb is to always maintain the original data as Mojo's processing power is way more complex and capable than any PC or mobile device.
 
DSD as a format has major problems with it; in particular it has two major and serious flaws:
 
1. Timing. The noise shapers used with DSD have severe timing errors. You can see this easily using Verilog simulations. If you use a step change transient (op is zero, then goes high) with a large signal, then do the same with a small signal, then you get major differences in the analogue output - the large signal has no delay, the small signal has a much larger delay. This is simply due to the noise shaper requiring time for the internal integrators to respond to the error. This amplitude related timing error is of the order of micro seconds and is very audible. Whenever there is a timing inaccuracy, the brain has problems making sense of the sound, and perceives the timing error has a softness to the transient; in short timing errors screw up the ability to hear the starting and stopping of notes.
 
2. Small signal accuracy. Noise shapers have problems with very small signals in that the 64 times 1 bit output (DSD 64) does not have enough innate resolution to accurately resolve small signals. What happens when small signals are not properly reproduced? You get a big degradation in the ability to perceive depth information, and this makes the sound flat with no layering of instruments in space. Now there is no limit to how accurate the noise shaper needs to be; with the noise shaper that is with Mojo I have 1000 times more small signal resolution than conventional DAC's - and against DSD 64 its 10,000 times more resolving power. This is why some many users have reported that Mojo has so much better space and sounds more 3D with better layering - and its mostly down to the resolving power of the pulse array noise shaper. This problem of depth perception is unlimited in the sense that to perfectly reproduce depth you need no limit to the resolving power of the noise shaper. 
 
So if you take a PCM signal and convert it to DSD you hear two problems - a softness to the sound, as you can no longer perceive the starting and stopping of notes; and a very flat sound-stage with no layering as the small signals are not reproduced accurately enough, so the brain can't use the very small signals that are used to give depth perception.
 
The second issue in using the transport to up-sample (44.1 to 176.4 say) is that the up-samplers in a PC or mobile device are very crude, with very limited processing power and poor algorithms. This results in timing problems, and like with DSD you can't hear the starting and stopping of notes correctly. These timing problems also screw up the perception of timbre (how bright or dark instruments sound), the pitch reproduction of bass (starting transients of bass lets you follow the bass tune), and of course stereo imagery (left right placement is handled by the brain using timing differences from the ears). Now Mojo has a very advanced algorithm (WTA) that is designed to maximise timing reconstruction (the missing timing information from one sample to the next) and huge processing power to more accurately calculate what the original analogue values are from one sample to the next. Its got 500 times more processing power than normal, and this allows much more accurate reconstruction of the original analogue signal.
 
So the long and the short is don't let the source mess with the signal (except perhaps with a good EQ program) and let Mojo deal with the original data, as Mojo is way more capable.
 
Rob

 
Also relevant:
 
Quote:
  It is always better to give Mojo bit perfect files and let Mojo do the work, as the processing within Mojo is much more complex and sophisticated than a mobile or PC.
 
So when you have an app that has a volume control, and no bit perfect setting, then set it to full volume on the app on the assumption that this will keep the data closer to the original file.
 
The volume control function on Mojo is much more sophisticated than the PC as I employ noise shaping and I do the function at a very high internal sample rate. Hopefully using the volume set to max on the app will mean the volume coefficient is 1.0000000... so it will return the original data.
 
Rob 

 
Quote:
  You can always do a listening test. If set to max against 50% say, and it hardens up (becomes brighter) with loud recordings then its clipping. If on the other hand the perception of sound-stage depth is reduced, then the volume control is degrading the sound.
 
If you do that test and can hear no difference then don't worry, its a good app volume control.
 
Rob
 
PS for fun I just did a very quick test using Dave. I listen to radio 3 using the BBC iPlayer. I normally have it set to max. I reduced the iPlayer volume control to half, boosted Dave volume control by +6dB - and yes I felt sound-stage depth was worse with lower iPlayer volume. 
 
 
 
 
 
So far I like using the Chord Mojo the most when listening to well-mastered vocalist stuff.
 
I think I get what reviewers and users mean when they remark that the soundstage seems narrower yet deeper when songs are played via the Mojo. I get a similar impression in the sense that the vocals seem more "forward". I don't mean sibilance or harshness, I mean the sense that the voice is being projected.
 
So I am really enjoying various CDs by vocalists. Happy to take a bit more time to now convert some of my fave vocalist albums to FLAC.
 
I also find the increase in detail and timing, especially when it comes to transients and drums, cymbals, hi-hats etc very impressive. But it's no something I want to emphasise too much because when listening to rock bands I wanna enjoy each song rather than concentrate on hearing the transients.
 
But really the Mojo is impressive for its price. Lots of times I get distracted cos I am like "wow! I never heard that detail before?!" and its a bit scary if I am walking on the street!

 
Perceived width is actually an aberration - so when image focus improves, the sensation of width diminishes. Its akin to looking at an image out of focus, then seeing it suddenly in focus - the size of the image gets smaller but you can see things much more accurately. Another way of looking at is perspective. If an instrument gets deeper into the sound stage, it naturally goes back and apparently decreases in width. So when you improve instrument placement focus in the sound stage, the perception of width will decrease.
 
There is an exception to this rule, in that you can encode sounds to sound wider than the loudspeakers, but this effect (replicating the phase delays and resonances of the earlobes and changing the left right phase) can increase width beyond the loudspeakers - it can also be used to encode height. But these effects are very rare and a bit hit and miss. Its these resonances and phase delays that allow binaural recordings on headphones to work. In this case, when image placement improves, then you get an increase in width and height - but as I said, these effects are very rarely found.
 
Rob

 
Also of interest:
 
  You can always do a listening test. If set to max against 50% say, and it hardens up (becomes brighter) with loud recordings then its clipping. If on the other hand the perception of sound-stage depth is reduced, then the volume control is degrading the sound.
 
If you do that test and can hear no difference then don't worry, its a good app volume control.
 
Rob
 
PS for fun I just did a very quick test using Dave. I listen to radio 3 using the BBC iPlayer. I normally have it set to max. I reduced the iPlayer volume control to half, boosted Dave volume control by +6dB - and yes I felt sound-stage depth was worse with lower iPlayer volume. 

 
 
  .... it is a 15T that is used on the Mojo.
 
That has 16,640 logic cells and 45 dsp cores. 44 cores are used in Mojo.
 
The overriding design decisions were about power consumption, so although more DSP cores are used than Hugo, that's to reduce power, as the DSP cores are run at a much lower clock speed. To give you another example of lower power, with Hugo when I needed a bigger multiplier I used one DSP core with FPGA fabric (logic cells) added to create the larger multiplier. With Mojo, to save power, I used multiple DSP cores and no fabric to create larger multipliers.
 
Only the WTA filter is different, the rest of the audio path has Hugo code.
 
Rob

 
Hugo and Dave don't use any kind of DAC chip, the analogue conversion is discrete using pulse array. The key benefit of pulse array - something I have not seen any other DAC technology achieve at all - is an analogue type distortion characteristic. By this I mean, as the signal gets smaller, the distortion gets smaller too. Indeed, I have posted before about Hugo's small signal performance - once you get to below -20 dBFS distortion disappears - no enharmonic, no harmonic distortion, and no noise floor modulation as the signal gets smaller. With Dave, it has even more remarkable performance - a noise floor that is measured at -180dB and is completely unchanged from 2.5v RMS output to no signal at all. And the benefit of an analogue character? Much smoother and more natural sound quality, with much better instrument separation and focus. Of course, some people like the sound of digital hardness - the aggression gets superficially confused with detail resolution - but it quickly tires with listening fatigue, and poor timbre variation, as all instruments sound hard, etched and up front. But if you like that sound, then fine, but its not for me.
 
On the digital filter front - original samples getting modified - actually the vast majority of FIR digital filters retain untouched the original samples, as they are known as half band filters. In this case, the coefficients are arranged so that one set is zero with one coefficient being 1, so the original sample is returned unchanged. The other set being used to create the new interpolated value. The key benefit of half band filters is that the computation is much easier, as nearly half the coefficients are zero, plus the filter can be folded so that the number of multiplications is a quarter of a non half band filter. When designing an audio DAC ASIC, the key part in terms of gate count is the multiplier, so reducing this gives a substantial improvement in die size, and hence cost. So traditional digital filters use a cascade of half band filters, each half band filter doubles up the oversampling - so a cascade of 3 half band filters will give you an 8 times over-sampled signal, with one sample being the unmodified original data. You can tell if the filter is like this as at FS/2 (22.05 kHz for CD) the attenuation is -6dB. The filters that are not like this are so called apodising filters, and my filter the WTA filter.
 
Going back eighteen years ago to the late 90's I was developing my own FIR filter using FPGA's. Initially, I was interested in increasing the FIR filter tap length as I knew from the mathematics of sampling theory that timing errors were reduced with increasing tap length. So the first test was to use half band Kaiser filters - going from 256 taps to 2048 taps gave an enormous sound quality improvement, so I had confirmed that tap length was indeed important subjectively. But at this point I was stuck; I knew that an infinite tap length filter with a sinc impulse response would return the original un-sampled signal perfectly - but the sinc function using only 16 bit accurate coefficients needs 1M tap FIR filter - and that would never happen, certainly not with 90's technology. So was it possible to improve the timing accuracy without using impossible tap lengths? After a lot of thinking and research, I thought there was a way - but it meant using a non half band filter, which would mean that the original sampled data would be modified. This was a big intellectual stumbling block - how can changing the original data be a good thing? But the trouble with audio is that neat simplistic ideas or preconceptions get in the way. Reality is always different, and reality can only be evaluated by a careful AB listening test. So I went ahead on this idea, and listened to the first WTA filter algorithm - and indeed it made a massive improvement in SQ - a 256 tap WTA sounded much better than 2048 tap half band Kaiser, even though the data is being modified. Why is this? The job of a DAC is NOT to reproduce the data it is given, but to reproduce the analogue signal before it is sampled. The WTA filter reconstructs the timing of the original transients much more accurately than using half band filters or filters that preserve the original data and it is timing of transients that is the most important SQ aspect.
 
So the moral of the tale? Don't let a simplistic technical story get in the way of enjoying music!                
  
 Rob

 
26,000 taps is the closest to a definitive statement as I've read ... the same as I've seen specified for Hugo.

I'm sure it's called out earlier in the thread, just going from memory as I'm far too lazy to search for it!

Actually it's about twice as many as Hugo but run at half the speed giving approximately the same number crunching power in terms of DSP ..... We have mentioned this before, but didn't want to put much focus on it as this is a small only part of the over design of Rob's overall topology

it was always our intention to try to match the performance of Hugo To do this without using as much power as Hugo. Therefore Rob used more DSP cores but run differently to match the performance of Hugo but at far lower power demands. JF

 
 
Quote:
  The earlier prototypes had less current than Hugo - 0.2A RMS. But last Feb we were with Nelson (Malaysian distributor) and I showed him the prototype. He brought some headphones, and we compared it to Hugo - and it was bad, a lot of clipping from Mojo, none from Hugo. It so happened (luckily) Nelson gave us possibly the only 8 ohm headphones on the market Final Audio Pandora.
 
So I upgraded the current to match Hugo, its the same 0.5A RMS. To do this I used 6 small OP transistors in parallel as size would not permit use of the large devices I use in Hugo.
 
Mojo development had some weird fortuitous events - the first time we showed the prototype happened to use the only 8 ohm headphones that would show problems - and the day we were deciding on the design Xilinx emailed me with a new FPGA that would enable Hugo performance but with the needed power, and happened to be in production just in time for Mojo. 
 
Quote:
   
1. The battery is capable of delivering 3A of current, and has very low impedance.
2. Mojo amplifier has a very high power supply rejection ratio.
3. The output is pure class A at 5v RMS into 300 ohm.
 
I just had an email today from a very experienced dealer that asked me this question:  
"Chord Mojo should have single amplification which drive to both headphone output, but if I'm using any headphones (HD800 for example, very heavy to drive) to put on headphone output 1 and I connect another headphone to headphone output 2 (Beyer T1 for example), I hear no differences on sound quality. Normally if one amplification section used to drive 2 headphones output, then once we connect the second headphone will make overall sound quality degradation (as the case of Beyerdynamic A20 amp, Grace Design M903, etc). What is the logical explanation of this? As it seems Mojo has 2 separate amplification sections which drive independently for each headphone output."
 
Of course Mojo does not have two amplification stages. It can drive two headphones with ease because it has exceptional low output impedance, and it has exceptional current linearity. So loading it with more headphones has no effect, unlike other headphone amplifiers.
 
Indeed, when I initially started designing Hugo I was shocked how poor from a measurement point of view headphone amps were. Poor output impedance, huge levels of distortion, and poor current linearity seem to be typical. And these things matter, if your goal is transparency and musicality. 
 
Rob
 
  I have just measured a Mojo into a 16 ohm load using an APX555 test equipment. With 1% THD 1 kHz single channel,  Mojo delivered 3.30 v RMS - that's 680 mW. Using 50 Hz, it was 668 mW RMS.
 
Rob

  I have done a quick measurement; with 30 ohms it is 4.25v RMS so that is 600 mW. For 50 ohms, I would expect 4.6v RMS or 423 mW RMS. This is with Mojo in blue battery, and at 1% THD with a continuous sine wave, power is RMS.
 
Rob

Quote:
  Into 300 ohms, fully charged battery, its 94 mW or 5.3v RMS at the 1% THD point.
 
Rob
 
 
Quote:
 
  Strange that the Hugo produces hiss (for those sensitive to it) with sensitive IEMs and the Mojo does not as the output specs are identical. If it were a lower powered Hugo aimed soley at IEM users, the lack of hiss would be expected, but it should be able to drive more demanding phones just as well. It is looking like a Hugo killer at a much lower price point.
The Hugo has a greater range of inputs and outputs and crossfeed for an additional £1000...

 
It's not strange if they used some kind of attenuator.

 
No attenuator as it would upset transparency.
 
I just reduced the noise - it is 125 dB dynamic range now. That said, "just reduced the noise" was not easy, it's one reason Mojo took so long to develop.
 
Rob
 
 
Quote:
 
Just a thought. Why do other DAC/headphone amps have amp sections when Hugo/Mojo get by without one, and many including Chord say it is more transparent? Have Chord got the patent for ampless amps :)


Because they can't using chip based DAC's. Chip DAC's have two current outputs. So you need two I to V converters (amps) then a differential to single ended amp, then a headphone buffer to deliver the current. You also need a lot of analogue filtering wrapped around these amps. So why are normal DAC's so complex in the analogue domain? Two reasons:
 
1. Silicon DAC's are horribly noisy, as the substrate and grounds are bouncing around due to switching activity. So to counter this, it is done differentially, which means the ground noise is cancelled. It also hides the problems of the reference circuitry, which can't be made with low enough impedance on silicon. This translates to more distortion, and crucially noise floor modulation.
 
2. Delta sigma converters run at low rates - best is at 12 MHz - this means that there is a lot of noise that must be aggressively filtered out in the analogue section. This also applies with R2R DAC's too as these have even worse problems due to the very slow switching speed.
 
So to run with a single amp section you need the DAC to be single ended and to run the noise shapers at much higher rates to reduce your filtering requirements. Because the analogue section with Mojo is discrete, I can use extremely low impedance and low noise reference supplies - something that is impossible on silicon. This has the other benefit of eliminating noise floor modulation (actually there is a lot more to it than this as there are countless other sources of noise floor modulation in a DAC). To make the filtering easier, the pulse array noise shapers run at 104MHz - over an order of magnitude faster than normal. There are other benefits to running the noise shapers at 104MHz, principally the resolving power of the noise shaper. Now soundstage depth is determined by how accurately small signals are reproduced. The problem with noise shaping is that small signals get lost - any signal below the noise shaper noise floor is lost information. But by running the noise shaper at much faster rates you solve this problem too - indeed Mojo noise shapers exceed 200dB THD and noise digital performance - that's a thousand times more resolving power than high end DAC's.
 
If I get time today I hope to publish noise floor modulation measurements showing Mojo has zero measured noise floor modulation. This level of performance does not happen on any other non pulse array DAC's at any price, and its the primary reason why Mojo sounds so smooth and musical.
 
Rob
 
  I have been seeing some comments describing Hugo as excellent DAC with a good headphone amp. Both comments, in my view, are wrong and way off the mark - and seeing these comments are starting to bug me, so I would like to get it off my chest. So forgive me if I am overstepping the mark - commenting on honest posts about a product I have designed, but I thought it might be useful for Head-fi'rs to read my views.
 
First, I would like to talk about what as a designer I am trying to accomplish, as it has a bearing on one's opinion of Hugo's sound. Imagine going around CES and carefully listening to all the high end hi-fi on show, so you can carefully listen to all the major high end brands available today. Next, listen center stage row 10 to an orchestra. Now, in my opinion, high end Hi-fi sounds from very bad to absolutely awful compared to live acoustic music. The key difference in the sound is variability - live acoustic music has unbelievable variations in the perception of space, timbre, dynamics and rhythm. Additionally, each instrument sounds separate and as distinct entities. By comparison, high-end audio is severely compressed - depth of sound stage is limited to a few feet (listen to off stage effects in say Mahler first - in a concert the off stage effects sound a couple of hundred feet away but on a hi-fi it is an ambient sound a few feet away). Timbre is compressed - you don't get a really rich and smooth instrument playing at the same time as something bright. The biggest problem is the dominance effect - the loudest instrument is the one that drags your attention away - this constant see-saw of attention is the biggest reason for listening fatigue, a major problem with Hi-fi.
 
So I am approaching designing of Hi-fi from the POV of accepting that there are enormous differences between conventional Hi-Fi and real music, and that I want my equipment to be as transparent as possible. Now some peoples idea of transparency is to use distortion to artificially enhance the sound, and this is a real problem with listening tests - a superficially brighter sound, giving the impression of better detail resolution, is often distortion. So a real challenge is defining what true transparency is. My definition, is to latch onto the idea of variations - if a modification makes the sound more variable, then its more expressive, and hence more transparent, even if it sounds, in tonal balance, darker or smoother and superficially less impressive. Now, if you think that your Hi-Fi sounds better than live acoustic music - then fine, we will agree to disagree. You are looking for a sculpted sound, not a truly transparent one, and I would strongly advise never to buy equipment designed by myself, as I am striving for equipment with no added sound.
 
So how does this relate to Hugo? Hugo was on the tail end of a long series of incremental improvements in digital design. I have spent the last 7 years on R and D to fundamentally improve aspects of DAC performance - improvements in the jitter rejection, RF noise filtering, noise shaper topologies, WTA filter length, analogue design plus a lot of other things. Moreover, Hugo took advantage of a big step forward in the capabilities of FPGA's - I could do important things that I knew influenced the sound but that previously were not possible due to FPGA limitations. So Hugo was at the confluence of two events - a big step forward from 7 years work in understanding digital design plus a major step forward in FPGA capability. It is just an accident that it happened with a portable headphone product.
 
So Hugo was the first instance when all these improvements came together. When I finally heard the pre-production unit with all the improvements in place I could not believe the sound quality improvements that I first heard. It completely changed my expectations of what was possible from digital audio - I was hearing things that I have never heard from Hi-fi ever - in other words, the gap from Hi-fi to live acoustic music was suddenly very much closer. Most notable was rapid rhythms being reproduced with breathtaking clarity - before piano music sounded like a jumble of notes, now I could hear each key being played distinctly. The next major change was timbre variations - suddenly each instrument had their own distinct timbre qualities, and the loudest instrument dominance effect was gone. Also gone was listening fatigue - I can listen for 12 hours quite happily.
 
But by far the biggest change was not sound quality, but on the musicality. I found myself listening and enjoying much more music, in a way I have never experienced before with a new design (and anybody who knows something of my designing career knows that is a lot of designs). 
 
So my conclusion is this: Hugo does things that no other DAC at any price point does. Now I can say readers saying, well OK he would say that anyway, it's his baby. True - I can't argue with that POV. But let's examine the facts:
 
1. The interpolation filter is key to recreating the amplitude and timing of the original recording. We know the ear/brain can resolve 4uS of timing - that is 250 kHz sampling rate. To recreate the original timing and amplitude perfectly, you need infinite tap lengths FIR filters. That is a mathematical certainty. Hugo has the largest tap length by far of any other production DAC available at any price.
 
2. RF noise has a major influence in sound quality, and digital DAC's create a lot of noise. Hugo has the most efficient digital filtering of any other production DAC - it filters with a 3 stage filter at 2048 FS. The noise shapers run at 104 MHz, some 20 times faster than all other DAC's (excepting my previous designs). What does this mean? RF noise at 1 MHz is 1000 times lower than all other DAC's, so noise floor modulation effects are dramatically reduced, giving a much smoother and more natural sound quality.
 
3. The lack of DAC RF OP noise means that the analogue section can be made radically simpler as the analogue filter requirements are smaller. Now in analogue terms, making it simpler, with everything else being constant, gives more transparency. You really can hear every solder joint, every passive component, and every active stage. Now Hugo has a single active stage - a very high performance op-amp with a discrete op-stage as a hybrid with a single global feedback path. This arrangement means that you have a single active stage, two resistors and two capacitors in the direct signal path -  and that is it. Note: there is no headphone drive. Normal high performance DAC's have 3 op-amp stages, followed by a separate headphone amp. So to conclude - Hugo's analogue path is not a simple couple of op-amps chucked together, it is fundamentally simpler than all other headphone amp solutions.
 
This brings me on to my biggest annoyance - the claim that Hugo's amp is merely good. Firstly, no body can possibly know how good the headphone amp in Hugo is, because there is not a separate headphone stage as such - its integrated into the DAC function directly. You can't remove the sound of the headphone amp from the sound of the DAC, it's one and the same.
 
Struck by these reports, I decided to investigate, as I see reported problems as a way of improving things in the future. I want to find weakness, my desire is to improve. So I tried loading the OP whilst listening on line level (set to 3v RMS). With 300 ohm, you can hear absolutely no change in sound. Running with 33 ohm, you can hear a small degradation - its slightly brighter. This is consistent with THD going from 0.0004% to 0.0007%. Note these distortion figures are way smaller than desktop headphone amps. Also note that with real headphones at this level you would be at typically ear deafening 115dB SPL. Plugging in real headphones (at much lower levels) gives no change in sound quality too. This has been reported by other posters - adding multiple headphones to Hugo does not degrade sound at all.
 
So how do we reconcile reports that desktop headphone amps sound better? I don't believe they do, its a case of altering the sound to suit somebody's taste. Now as I said at the beginning of this post, that is not what I want to do - I want things to sound transparent, so that we can get closer to the sound of live acoustic music. Adding an extra headphone amp will only make things worse as extra components degrades transparency. Another possibility is that people are responding against Hugo's unusually (for a headphone amp) low output impedance of 0.075 ohms. Now, compared to headphone amps of 2 to 33 ohms impedance, this will make the sound much leaner with less bass. Additionally, the improvements in damping can be heard as a much tighter bass with a faster tempo. So if you find your headphone too lean, the problem is not Hugo's drive - your headphone is just been driven correctly.                 
 
Just to close to all Hugo owners - enjoy! I hope you get as much fun from your music as I have done with Hugo. 

 
 

 
 
Quote:
 
  I found the reverse.  I'm using Sennheiser HD-25 1 II: directly out of the Mojo the sound seems present and correct, but when used with a Ray Samuels SR-71a, the sound goes to a whole new level.  The sound becomes rock solid and more like listening to musicians playing instruments; without the Ray Samuels the sound seem to collapse in on itself and become more hi-fi (ie impressive noises but less music).  To my ears, the extra amplification is not adding tonal euphony but is instead making the most of the DAC.
 
I have a theory that it's to do with the power supply: when using headphones more current is drawn and in a varying manner, ie it varies with the music.  This varying of current affects (I think modulates) the power supply voltage which affects the DAC, amplification and ultimately the sound. By connecting directly to an amp, there is less current drawn and no variation.  This might also explain why companies such as Naim claim improvements to their amps' sound quality when external power supplies are added.  Just my 2CW.


I do not buy this all. You need to bear in mind several facts:
 
1. The battery is capable of delivering 3A of current, and has very low impedance.
2. Mojo amplifier has a very high power supply rejection ratio.
3. The output is pure class A at 5v RMS into 300 ohm.
4. Reducing the output load only starts to increase distortion with 33 ohms - at this level it is very much lower than other headphone amps. The HD25 is a very easy 70 ohms.
5. Mojo is designed to drive loudspeakers. You will be amazed hearing it fill the room with beautiful sound using efficient 8 ohm horn loudspeakers.
 
Some people like the sound of more distortion - 2nd harmonic fattens the sound making everything sound phat, soft and rounded. But its not natural, nor do I find it musical, as everything sounds phat. I want soft sounds to sound soft, and sharp sounds to sound sharp - not everything to have a soft sheen on things all the time.
 
I can give you another example. I just had an email today from a very experienced dealer that asked me this question:
 
"Chord Mojo should have single amplification which drive to both headphone output, but if I'm using any headphones (HD800 for example, very heavy to drive) to put on headphone output 1 and I connect another headphone to headphone output 2 (Beyer T1 for example), I hear no differences on sound quality. Normally if one amplification section used to drive 2 headphones output, then once we connect the second headphone will make overall sound quality degradation (as the case of Beyerdynamic A20 amp, Grace Design M903, etc). What is the logical explanation of this? As it seems Mojo has 2 separate amplification sections which drive independently for each headphone output."
 
Of course Mojo does not have two amplification stages. It can drive two headphones with ease because it has exceptional low output impedance, and it has exceptional current linearity. So loading it with more headphones has no effect, unlike other headphone amplifiers.
 
Indeed, when I initially started designing Hugo I was shocked how poor from a measurement point of view headphone amps were. Poor output impedance, huge levels of distortion, and poor current linearity seem to be typical. And these things matter, if your goal is transparency and musicality. 
 
Rob
 
  There has been some talk about Mojo's hiss when silent. We publish the noise output voltage and its 3uV - that's same as an iPhone, and a little bit better than an AK240. With the Shure SE846 (pretty much the most sensitive IEM you can get) the 3uV translates to a noise of 24 dB SPL - and would be the same as the AK240 and the iPhone - but - and this is a big point - Mojo will also deliver over 5V RMS with the noise at 3uV still.
 
24dB will be audible to some, and not to others, as you naturally hear hiss with IEMs stuck in your ears. My K10's are completely inaudible with Mojo powered or not powered, similarly the ultimate ears UERM. But these devices work out at 6 dB SPL as they are sensibly sensitive.
 
Rob

  Just to reiterate on the hiss issue - Mojo has only 3uV of residual noise (that's the level with no signal). I have not seen a DAC, DAP, or mobile phone that betters this number, and this will determine the hiss level you hear. With sensible sensitivity IEM's (Noble, Ultimate ears, Dita) I can hear absolutely no added hiss from Mojo - that is turning Mojo on or off has no change in hiss levels from normal background noise.
 
Rob
 
  1. How does PSU design influence the sound quality of Rob's DACs?
 
  1. Detailed Blog on Listening Tests
 
 

 
 
(NB: please also view the VIDEOS section!)
 
 
 
www.the-ear.net/how-to/rob-watts-chord-mojo-tech
 
www.youtube.com/watch?v=3e7SRXP3RHI
 
Quote:
x RELIC x said:
/img/forum/go_quote.gif
  Q&A with Rob Watts:
 
 
Q: Were you able to fit the same tap filter length as the Hugo (26, 384 taps) with the same WTA filter in the Mojo?
 
A: "Mojo shares an extremely similar code as Hugo - the only change is the WTA filter is redesigned to accommodate 768 kHz. The new filter is broadly equivalent apart from this."
(Comment): When I pushed the tap length question with Chord they replied that "it will be a good while in the future before they publish this information, if at all". "The implementation within Mojo is different, but it’s not inferior to anything that we’ve done".
 

 

Q: In the Mojo presentation draft it mentions “Hugo like sound quality and musicality”. What differences in audio presentation would you say the Mojo has compared to the Hugo?
 
A: "Bearing in mind it’s use I have optimized the noise performance in order to make it sound smoother."
 

 

Q: The design for the Mojo began in 2012. Is it safe to say the Mojo R&D led to the Hugo until the technology caught up for the Mojo’s design target? Or, were they completely separate design goals?
 
A: “The R&D of Hugo and Mojo ran in parallel - the very first prototype (2012) was more like Mojo, then work switched to Hugo. Then I worked on Mojo in the background, with development getting really busy starting in Nov 2014. We built over 50 prototypes, as I had a lot of issues to contend with - thermals, charging, and getting SQ to be identical when charging were major headaches."
 

 

Q: Does the Mojo deal with jitter with the same DPLL as the Hugo?
 
A: “Yes, the DPLL is identical."
 

 

Q: I see the Mojo has an even better THD spec than the Hugo.
 
A: “Lower noise means better measurements."
 

 

Q: Is the Mojo analogue section Class A like the Hugo?
 
A: “The actual OP stage is identical - same OP transistor silicon - but I used 6 small transistors in parallel rather than 3 large devices. It’s biased at the same Class A level."
 

 

Q: Does Mojo have cross feed?
 
A: “No cross feed.”

 
 
Also see: Munkonggadgets interview with John Franks
 

 
Related discussion (not Mojo-specific, but much of it does apply to Mojo) on page 56 of this Rob Watts interview
 
...and this John Franks interview may be of interest, too.
 
Also, this one
 
 
 
and this: Interesting historical background of Rob's DAC design approach (video interview with Rob Watts & John Franks) (well worth watching)
 

 
 
  1. ohm-image.net/data/audio/rmaa-chord-mojo-24-bit
 
  1. goldenears.net/board/5904087
 
  1. stereophile.com/content/chord-electronics-mojo-da-headphone-amplifier-measurements
 
  1. hi-fiworld.co.uk/index.php/cd-dvd-blu-ray/62-cd-reviews/776-chord-mojo-review
 
 
 
  1. Quote:
  I promised some time ago that I would show some measurements showing Mojo's performance. My reasoning for this was that Mojo does things that no other (non Chord) DAC does at any price; I was kind of annoyed that some people were comparing it to $100 DACs when the true competitors were $100K - and I kind of get that, its difficult to take Mojo seriously given its size and price. But if you could see the design complexity that goes inside Mojo then one could appreciate how much better it is; it really is vastly more complex than other DAC's, and this complexity is needed to recreate the original analogue signal accurately.
 
But I can show you that something special is going on from measurements. Take a look at this plot. This is a FFT of a 1kHz output at 2.5v RMS into a 300 ohm load (blue trace) and then with no signal (red trace):
 

 
Now what is very interesting is the noise floor at -175dB - it does not change at all with 2.5v or nothing which indicates a complete absence of measurable noise floor modulation. Noise floor modulation is extremely important subjectively - you perceive the slightest amount as a brightness or hardness to the sound. When it gets bad, you hear glare or grain in the treble. All DAC's (apart from Chord DAC's) suffer from measurable noise floor modulation - typically the noise floor would be -160 dB with no signal, and -140 dB at 2.5v RMS. Some Class D amps are awful with noise floor at -120 dB (one reason why Class D often sounds so bad).
 
To get this measurement is a massive challenge, as ADC's themselves have large amounts of noise floor modulation, way bigger than my DAC's. The only test instrument that has noise floor modulation that can actually measure Mojo's performance is the APX555. This uses a novel approach to solving the issue - 4 ADC's and an analogue notch filters. The outputs are combined in the digital domain, so this means one ADC is handling the fundamental sine wave, another ADC looks at the noise via the notch filter. So you will only be able to measure Mojo's true performance using the APX555. 
 
Many posters have commented on how smooth and musical sounding Mojo is - and its in part down to the absence of measurable noise floor modulation. Actually getting this performance is very complicated, as within the DAC there are a enormous number of mechanisms to create noise floor modulation. One reason why its taken me 20 years of DAC development to do it!
 
Rob
 

 
 
(NB: please also view the VIDEOS section!)
 
 
  1. forbes.com/sites/marksparrow/2016/07/11/hi-res-audio-can-be-in-the-palm-of-your-hand-with-chords-mojo-dac-for-smartphone-users/#6b4011055255
 
  1. dailymail.co.uk/home/event/article-3690447/Chord-Mojo-flashback-days-British-hi-fi-single-best-audio-upgrade-buy.html
 
  1. whathifi.com/chord/mojo/review  (also see: whathifi.com/news/chord-electronics-dominates-best-dacs-2015)
 
  1. cnet.com/uk/news/chord-mojo-maximum-sound-quality-from-a-tiny-digital-converterheadphone-amplifier
 
  1. stereophile.com/content/chord-electronics-mojo-da-headphone-amplifier#m76BmGJUB2rti7gI.97
 
  1. telegraph.co.uk/luxury/technology/92513/chords-exceptional-audio-mojo.html (Ken Kessler)
 
  1. alphr.com/audio/1003966/chord-mojo-review-make-your-smartphone-sound-amazing
 
  1. metal-fi.com/chord-electronics-mojo/ (this contains quite an interesting discussion)
 
  1. the-ear.net/review-hardware/chord-electronics-mojo-portable-dacheadphone-amp
 
  1. audiovideo.fi/testi/chord-mojo-da-muunnin-kuulokevahvistin-testissa (in Finnish)
 
  1. stereo.net.au/reviews/review-chord-electronics-mojo-headphone-amplifier-dac
 
  1. artsexcellence.com/downloads/reviews/chord.mojo.artsexcellence.english.pd
 
  1. hifiplus.com/articles/chord-electronics-mojo-portable-dacheadphone-amp
 
  1. headphone.guru/the-chord-mojo-the-amazing-599-00-portable-wonder
 
  1. avforums.com/review/chord-mojo-dac-headphone-amp-review.12008
 
  1. digitaltrends.com/home-theater/chord-mojo-dac-amp-hands-on
 
  1. moon-audio.com/chord-mojo-dac-headphone-amp.html
 
  1. headfonics.com/2016/04/the-mojo-by-chord-electronics
 
  1. hi-fiworld.co.uk/index.php/cd-dvd-blu-ray/62/776.html
 
  1. howtospendit.ft.com/audiovisual/109351-chord-mojo
 
  1. blog.son-video.com/en/2016/09/review-chord-mojo
 
  1. headfonia.com/review-chord-mojo-the-chosen-one
 
  1. headfonia.com/review-chord-mojo-hot-or-not
 
  1. headphonescanada.ca/blog/chord-mojo-review
 
  1. head-fi.org/products/chord-mojo/reviews/14867
 
  1. head-fi.org/t/784602/10995#post_12328398
 
  1. head-fi.org/t/784602/5850#post_12110107
 
  1. head-fi.org/t/784655/chord-mojo-review
 
  1. custom-cable.co.uk/blog/chord-mojo-review
 
 
 
  1. "Now I finally understand the name. Chord’s got Mojo! Every audiophile needs his fix, and with Mojo he can get it anywhere."
 
  1. "A defining moment in audio reproduction. Very real holographic, 3d sound. The capabilities of this are literally awesome."
 
  1. "Chord Mojo Review - The Game Really Has Changed!" (NB: this review includes a useful short Q&A with Rob Watts)
 
  1. "A class leading sound quality DAC/AMP in a tiny footprint that work well with wide ranges of headphones."
 
  1. "The Chord Mojo is easily one of the most, if not the most outstanding product of 2015."
 
  1. "Mojo is very good at what it is intended to be used as – a portable DAC/amp"
 
  1. "Desktop Capable and Wholly Portable, A swiss Army knife in all but name!"
 
  1. "Mojo on the Go: A Review of Mojo and It's use from a Portable Perspective"
 
  1. "Great Value for Money. Possible Consideration for an All in One Solution."
 
  1. "Very musical. Great with FIIO digital output and USB output from a PC"
 
  1. "Chord Mojo : A small, affordable and highly musical portable device"
 
  1. "Absolutely the best portable amp/DAC combo on the market"
 
  1. "Chord Mojo: Top-notch sound in a small, portable device!"
 
  1. "Fulfillment of Foolish and Overwrought Expectations"
 
  1. "Excellent Mojo, A Unicorn at this price/performance"
 
  1. "The Chord Mojo: A Budget-Minded Rookie's Take"
 
  1. "A very accurate DAC/Headphone Amplifier"
 
  1. "CHORD Mojo - DAC/Headphone Amplifier"
 
  1. "A Lamborghini for the price of a Porsche"
 
  1. "Mojo Brings the Best Out of My IEMs"
 
  1. "MOJO: a little gem, highly musical"
 
  1. "Amazing sounding all in one unit!"
 
  1. "The Mojo. Get Inside Your Music"
 
  1. "GREAT HIGH END PRODUCT!!"
 
  1. "A great little portable device"
 
  1. "Audiophile Basshead grade"
 
  1. "The Magical Black Box"
 
  1. "Big Bang, Little Box"
 
  1. "An Instant Classic"
 
  1. "I found my MoJo"
 
  1. "Excellent dac"
 
  1. "Chord Mojo"
 
 

 
"We use it all the time..even for testing internally. It sounds awesome."
 
www.head-fi.org/t/784602/20595#post_12753517
 
www.head-fi.org/t/784602/26865#post_13061577
 
www.head-fi.org/t/784602/15165#post_12472084
 
www.head-fi.org/t/784602/15390#post_12480734
 
www.head-fi.org/t/784602/11055#post_12330962
 
www.head-fi.org/t/784602/16260#post_12516947
 
www.head-fi.org/t/784602/19365#post_12691852
 
www.head-fi.org/t/784602/18495#post_12634409
 
www.head-fi.org/t/784618/chord-mojo-impressions-thread
 
www.head-fi.org/t/784602/12105#post_12381029 (also discusses an innovative stacking approach)
 
www.head-fi.org/t/784602/13620#post_12430618
 
www.head-fi.org/t/784602/1140#post_12007799
 
www.head-fi.org/t/784602/15630#post_12492483
 
www.head-fi.org/t/784602/11775#post_12363801
 
www.head-fi.org/t/784602/1185#post_12008368
 
www.head-fi.org/t/784602/6825#post_12161630
 
www.head-fi.org/t/784602/1440#post_12011978
 
www.head-fi.org/t/784602/1470#post_12012541
 
www.head-fi.org/t/784602/1650#post_12015292
 
www.head-fi.org/t/784602/1755#post_12016908
 
blog.moon-audio.com/chord-mojo-review/
 
www.digitalaudioreview.net/2015/10/chord-electronics-mojo-portable-audios-new-talisman/
 
gavinsgadgets.com/2015/10/23/the-chord-mojo-the-game-changer-has-arrived-first-impressions/
 
www.hifiplus.com/articles/first-look-chord-mojo-portable-dacheadphone-amp/
 
www.digitalaudioreview.net/2016/02/chord-electronics-mojo-lost-found/
 
www.head-fi.org/t/784602/1950#post_12019292
 
www.head-fi.org/t/784602/13110#post_12417346
 
www.head-fi.org/t/784602/2025#post_12020114
 
www.head-fi.org/t/784602/2100#post_12021153
 
www.head-fi.org/t/784602/2535#post_12028236
 
www.head-fi.org/t/784602/3165#post_12038719
 
www.head-fi.org/t/784602/3360#post_12041678
 
www.head-fi.org/t/784602/3675#post_12048000
 
www.head-fi.org/t/784602/3855#post_12050914
 
www.head-fi.org/t/784602/4620#post_12065951
 
www.head-fi.org/t/784602/4665#post_12067288
 
www.head-fi.org/t/784602/4740#post_12068725
 
www.head-fi.org/t/784602/4800#post_12070256
 
www.head-fi.org/t/784602/4815#post_12070908
 
www.head-fi.org/t/784602/4815#post_12070955
 
www.head-fi.org/t/784602/5010#post_12077101
 
www.head-fi.org/t/784602/5475#post_12094314
 
www.head-fi.org/t/784602/5565#post_12096634
 
www.head-fi.org/t/784602/5805#post_12106201
 
www.head-fi.org/t/784602/5775#post_12103485
 
www.head-fi.org/t/784602/5820#post_12106718
 
www.head-fi.org/t/784602/5835#post_12107978
 
www.head-fi.org/t/784602/5880#post_12111500
 
www.head-fi.org/t/784602/5880#post_12112325
 
www.head-fi.org/t/784602/5970#post_12116364
 
www.head-fi.org/t/739712/1964-ears-adel-iems/2835#post_12137561
 
www.head-fi.org/t/784602/6210#post_12131983
 
www.head-fi.org/t/784602/6450#post_12145215
 
www.head-fi.org/t/784602/6615#post_12151064
 
www.head-fi.org/t/784602/6705#post_12154267
 
www.head-fi.org/t/784602/6720#post_12155146
 
www.head-fi.org/t/784602/6735#post_12155440
 
www.head-fi.org/t/784602/15960#post_12505605
 
www.head-fi.org/t/784602/7035#post_12172329
 
www.head-fi.org/t/784602/7050#post_12172799
 
www.head-fi.org/t/784602/7050#post_12173127
 
www.head-fi.org/t/784602/7095#post_12173591
 
www.head-fi.org/t/784602/7455#post_12187762
 
www.head-fi.org/t/784602/7890#post_12203042
 
www.head-fi.org/t/784602/7905#post_12203566
 
www.head-fi.org/t/784602/9570#post_12268684
 
www.head-fi.org/t/784602/9660#post_12272381
 
www.head-fi.org/t/784602/9690#post_12273665
 
www.head-fi.org/t/784602/9690#post_12273890
 
www.head-fi.org/t/784602/9780#post_12276415
 
www.head-fi.org/t/784602/10095#post_12291355
 
www.head-fi.org/t/784602/10575#post_12309334
 
www.head-fi.org/t/784602/10650#post_12313993
 
www.head-fi.org/t/784602/10650#post_12314170
 
www.head-fi.org/t/784602/11115#post_12333070
 
www.head-fi.org/t/784602/11775#post_12363723
 
www.head-fi.org/t/784602/11820#post_12366263
 
www.head-fi.org/t/784602/12675#post_12404483
 
www.head-fi.org/t/784602/12705#post_12405102
 
www.head-fi.org/t/784602/12795#post_12407006
 
www.head-fi.org/t/784602/12795#post_12407032
 
www.head-fi.org/t/784602/14430#post_12452128
 
 

 
 

Other relevant posts and threads:

 
 
Just a thought. Why do other DAC/headphone amps have amp sections when Hugo/Mojo get by without one, and many including Chord say it is more transparent? Have Chord got the patent for ampless amps :)


Because they can't using chip based DAC's. Chip DAC's have two current outputs. So you need two I to V converters (amps) then a differential to single ended amp, then a headphone buffer to deliver the current. You also need a lot of analogue filtering wrapped around these amps. So why are normal DAC's so complex in the analogue domain? Two reasons:
 
1. Silicon DAC's are horribly noisy, as the substrate and grounds are bouncing around due to switching activity. So to counter this, it is done differentially, which means the ground noise is cancelled. It also hides the problems of the reference circuitry, which can't be made with low enough impedance on silicon. This translates to more distortion, and crucially noise floor modulation.
 
2. Delta sigma converters run at low rates - best is at 12 MHz - this means that there is a lot of noise that must be aggressively filtered out in the analogue section. This also applies with R2R DAC's too as these have even worse problems due to the very slow switching speed.
 
So to run with a single amp section you need the DAC to be single ended and to run the noise shapers at much higher rates to reduce your filtering requirements. Because the analogue section with Mojo is discrete, I can use extremely low impedance and low noise reference supplies - something that is impossible on silicon. This has the other benefit of eliminating noise floor modulation (actually there is a lot more to it than this as there are countless other sources of noise floor modulation in a DAC). To make the filtering easier, the pulse array noise shapers run at 104MHz - over an order of magnitude faster than normal. There are other benefits to running the noise shapers at 104MHz, principally the resolving power of the noise shaper. Now soundstage depth is determined by how accurately small signals are reproduced. The problem with noise shaping is that small signals get lost - any signal below the noise shaper noise floor is lost information. But by running the noise shaper at much faster rates you solve this problem too - indeed Mojo noise shapers exceed 200dB THD and noise digital performance - that's a thousand times more resolving power than high end DAC's.
 
Mojo has zero measured noise floor modulation. This level of performance does not happen on any other non pulse array DAC's at any price, and its the primary reason why Mojo sounds so smooth and musical.
 
Rob

 
Quote:
  the OP stage is integrated with the OP filter. This means that Mojo analogue section is very simple, so giving Mojo's transparency, but the downside is a small variation in frequency response with load impedance.
 
Rob
 
Quote:
 
  .... the Mojo on line level mode - does this still run thru the Mojo's amp? from how i understand your earlier descriptions, buth the amping and DAC is done in the FPGA?
thus there is no way to truly use it as a dac without double amping?

Line level mode is just a volume preset for the volume control - nothing else changes.
 
Mojo has an FPGA (which is digital logic only) a discrete DAC (turning digital signals to analogue via flip-flops and resistors) and a single output amplifier - and that is it.
 
Conventional DAC headphone amps use differential outputs and have two I to V converters (current to voltage), a differential to single ended converter, and an output amplifier. Wrapped up with that is a analogue filter. So that's a lot of passive components and four amplifiers in the signal path. 
 
Because Mojo's FPGA has extensive digital filtering (at 2048 FS) and has a noise shaper that runs at a very high rate (104MHz) and uses a discrete DAC, I can keep the analogue section radically simpler, and this is one reason why Mojo is so transparent compared to all other DAC amps.
 
Rob

 
  @xtr4 i understand the FPGA designs makes the dac and amp essentally the same... what im really trying to get at is, can the FPGA's amp functions be bypassed so it is used simply as a DAC, and the two 3.5mm outs are true line outputs to prevent double amping
Paste

No, you need at least one amplifier to do the critical I to V conversion. Now it is possible to design a voltage only DAC (no amp at all), but they sound poor due to lots of problems - the largest being the huge amount of distortion you get doing it that way. Believe me, if I could make it simpler I would. The key that Mojo has is extremely low distortion and noise (0.00017% 3V 300 Ohms) but only one single amplifier in the signal path - and this amp combines headphone drive, filtering and I to V conversion in a single stage.
 
Rob

 


 
 
 
Of course the balanced output is going to be better than the Mojo, the Mojo doesn't have balanced output.

No that simply is not correct! A single ended design, done right with a large enough voltage swing will easily out perform a balanced output. Balanced designs are used by some designers to overcome inherent limitations within designs. Usually to overcome substrate noise on the chip that shouldn't be there or to increase the output voltage swing of their amplifiers. We don't suffer those limitation or problems so we don't need a dodgy fix for them. Our measurements clearly show this. Sorry to burst you bubble man.

Balance operation is a fix for problems we don't have. We have no substrate noise and we have plenty of output swing. Single ended done right is far better than a balanced design far less distortion.

 
Quote:
Originally Posted by Rob Watts /img/forum/go_quote.gif
Quote:
Originally Posted by agisthos /img/forum/go_quote.gif
Rob you should give a definitive 'why SE is better' explanation. Get it over with, because many (most) audiophiles have been biased towards balanced and are not going to understand where you are coming from.
 
One good argument I heard from the Densen founder (Thomas Sillesen) is that each half of the signwave runs through a series of components that will always have tolerances different from each other, so when combining the signal they will not ever match, causing an increase in distortion (of some kind I cannot remember).
 
Charles Hanson, of Ayre, who is a proponent of fully balanced equipment, has even stated that for pure sound quality SE will always sound better, but this is on the bench, where the power supply and analog signal stages can be kept physically apart. When putting them in a box he prefers balanced.

Well this is a complex subject, and sometimes a balanced connection does sound better than single ended (SE) - in a pre-power context - but it depends upon the environment, and the pre and power and the interconnect. But the downside of balanced is that you are doubling the number of analogue components in the direct signal path, and this degrades transparency. In my experience every passive component is audible, every metal to metal interface (including solder joints - I once had a lot of fun listening to solder) has an impact - in case of metal/metal interfaces it degrades detail resolution and the perception of depth. So going balanced will have a cost in transparency.
 
In DAC design, going balanced is essential with silicon design; there is simply too much substrate noise and other effects not too. But with discrete DAC's you do not need to worry about this, so going SE on a discrete DAC is possible, and is how all my DAC's are done. But differential operation hides certain problems (notably reference circuit) that has serious SQ effects; so going SE means those problems are exposed, which forces one to solve the issue fundamentally. In short, to make SE work you have to solve many more problems, but the result of solving those problems solves SQ issues than differential operation hides when you do measurements.
 
Rob 

Quote:
Originally Posted by Rob Watts /img/forum/go_quote.gif
Component count is very important for transparency. Doubling the number of parts in the direct signal path does degrade depth perception and detail resolution.
 
But there is another problem with balanced operation. Imagine a balanced differential in, differential out amplifier. The input stage is normally a differential pair (maybe cascoded) with a constant current source. Now the input stage is free to move up and down to accommodate the common mode voltage - but the input stage common mode impedance is non linear, and if the common mode voltage has a signal component (it always will have due to component tolerances) then this will create a signal dependent error current, thereby generating distortion. Unfortunately, the negative feedback loop of the amplifier can't correct for this distortion as it can't see the error on the summing nodes. So there will always be a limit to the performance. With SE operation, this problem does not occur, as the differential input stage is clamped to ground.
 
Now DAC designers are well aware of this - that's why all high performance DAC's use two single ended I to V converters from the current OP of the DAC's, then use a differential to SE converter to create the voltage OP. There are other reasons for doing this as well, as the DAC requires a very low impedance virtual ground for low distortion, and you can only get this using dual SE amps - another problem is RF and its much easier to decouple SE than differentially - this in turn creates a lot more noise floor modulation, making it sound less smooth.
 
But for me the most important is transparency. I had an amp that had two modes - differential or SE - listening in balanced mode flattened the sound stage depth dramatically,and it sounded harder, less smooth. That said, there are circumstances when balanced operation can be better than SE, for example when you are looking at connecting a pre-amp to a power amp, and what is best depends upon particular circumstances. In short, if SE operation is noisy, try balanced.
 
Rob 

 
 
 
 
@robwatts @mojo ideas

How about an impedance module that allowed us to adjust output impedance until we perfectly matched mojo to our ciems/headphones?

The technically perfect impedance is zero, and that's why I worked so hard to get it as low as 0.075 ohms with Mojo.
 
The reasons going for as close as zero are:
 
1. Frequency response. The impedance of the headphone varies with frequency, and so by having a high output impedance will cause frequency response variations. Zero impedance eliminates this problem.
 
2. Distortion. The impedance of a headphone varies with level, and having a higher output impedance will increase the total distortion - given that Mojo distortion is so low, this is actually quite a significant an effect. Again, zero impedance eliminates this problem.
 
3. Damping factor - probably the most important reason. A drive unit is a resonant system - that is a mass on a spring - that is damped mechanically and electrically. Electrical damping is due to the headphone creating a current due to the motion of the driver in the magnetic field - and how well this is controlled depends on the electrical impedance the driver sees - in our case, the cable impedance and Mojo's impedance. Again, zero impedance gives the best damping, with an infinite damping factor.
 
I did some listening tests many years ago with loudspeakers and damping factor and found that it made a massive difference to the sound. Damping of 10 gave a very soft, big fat bass - but everything sounding one note in the bass - simply because the loudspeaker was doing its own thing at the resonant frequency. Going from 10 to 100 gave a tighter bass, with much better pitch reproduction - you could follow the bass line much more easily. Above 100 to 1000 it sounded tighter - no big change in pitch (being able to follow the bass tune) but the perceived tempo of the music became faster as transients are much better controlled. Going above 1000 gave a small improvement in how tight it sounded.
 
Rob

 
 
 
Please note that the following quote was posted in the 2qute DAC thread, and is referring to the Hugo, so please exercise some discretion in that the Hugo is not 100% identical to the Mojo, but the majority of this information does apply equally-well to Mojo:
 
 
  Dear Rob
 
What is a OP stage? I understand discrete stage is better than op-amp, could you explain why? As I understand the Hugo has no analog volume control, so the output from the DAC doesn't go through a preamp (like one of the competing products from Salisbury)
 
Also what is a pulse array dac? is it similar to Delta Sigma or the resistor ladder Dac?  Is the sound of the hugo due to the filter or due to filter/dac combination? Also if you were to use this filter with a conventional resistor ladder DAC would it work?
 
Thanks
Analog

Welcome to Head-Fi analogmusic, and I am pleased you are enjoying more musicality from your music with Hugo - which is what this is all about!
 
What is an OP stage?
OP is output, and it replaces rather poor OP stages within op-amps. When faced with designing the electronics of Hugo, I had no experience of designing headphone amps - looking into devices that supplied headphones, they were very poor. So I designed it as if it was a power amp (I've designed lots of those) and gave Hugo the ability to drive 8 ohm loudspeakers directly - which means lots of current - in Hugo's case I set it too 0.5A RMS. You will not get this current from op-amps or headphone drive chips, so I had to design a discrete amp. Now to get the best transparency there needs to be a single feedback path, so the discrete OP stage needs to be within the op-amp's global feedback path. Since the op-amps are very high gain bandwidth product devices (high speed), that meant designing a Class A OP stage with very low propagation delay, so that the circuit would remain stable. Now the op-stages in op-amps are pretty poor to awful, so when I got the first prototype I was very pleased at how good the OP stage sounded, and how much lower distortion was (particularly high order harmonics) - even when using the op-stage in DAC mode with easy loads. Indeed, I now use this arrangement all the time now, as it really improves the performance of the op-amp - that's why 2 Qute has it too. The OP stage is by far the weakest part of all op-amps and this is simply because one can use a decent Class A bias current, and very substantial OP transistors, so thermal stability is ensured. And yes, Hugo does not have an analogue volume control, so this means the analogue section is very simple (just 2 resistors and capacitors in the direct signal path). Simple analogue gives much more transparency.
 
What is a Pulse Array DAC?
This is not an easy answer, as its complex and of course proprietary. But firstly the history. I first started designing DAC's in 1989, when the first delta-sigma bitstream devices from Phillips came out - these were DSD 256 DAC's (or PDM dac's). Now they were quite musical, but had technical and SQ problems - but they had very good low signal performance, and analogue distortion characteristic (small distortion for small signals unlike R2R DAC's which have more distortion for small signals due to glitch energy and resistor matching problems - issues that are impossible to solve). The biggest problem was limiting of resolution - unlike PCM, where ultra small signals are buried in the dither and so perfectly preserved, with delta-sigma the noise floor is a cliff edge for low level signals - any small signal below the resolving power of the noise shaper is lost forever. To overcome this, I used 8 PDM noise shapers with different dither, and summed the output in the analogue current to voltage converter (I to V). This gave much better performance, but I knew that much more was possible. So I started creating my own noise shapers and DAC technology using FPGA that were just becoming available (1994 now). What I needed was much higher resolution so the noise shaper OP is 5 bits not 1 bit, and I ran the noise shapers at a much higher rate - 2048 times not 256 times. Running at a faster rate means that you have more permutations of OP, which translates to much better performance. Run a 5th order noise shaper at ten times the speed, you can get in the digital domain, up to 100 dB lower distortion and noise - that's a 100 dB improvement in small signal resolution, so running at much higher rates gives massive improvements in SQ and measurements. Twenty years on, and I am still the only silicon/FPGA DAC designer running as high as this rate - delta-sigma DAC's are still stuck at 256 times or below.
 
But changing from single bit to multi-bit noise shaping may throw the baby out with the bathwater. The primary benefit of single bit is that it can (if you are very very careful) have zero small signal distortion, as there are no resistors to balance, as there is only one. With 16e Pulse Array, there are 16 PWM elements, and each element has on the long term exactly the same data, but instantaneously slightly different data. The benefit of the Pulse Array scheme is that when the elements are slightly different in value, it creates a fixed signal independent noise, and absolutely no distortion, but has innately higher resolution of 5 bits. That's why Hugo has (uniquely compared to other non Pulse Array DAC's) no measurable distortion, or any other artifact, for signals below -30 dBFS (see plots in previous posts). Additionally, because of the way the array is composed, master clock jitter has no significant affect - random jitter gives a tiny insignificant fixed noise. Its why I don't go endlessly on about femto clocks as the DAC is innately jitter insensitive. There are many more problems with noise shaping, as it is a very complex subject, but this will give you a flavour of the issues involved.  
 
Is the sound of the hugo due to the filter or due to filter/dac combination?
The sound of Hugo is down to lots of things, but of course the primary problem that Hugo addresses is the time domain one. That's where we are converting the sampled data into the original un-sampled continuous analogue waveform - the original signal at the ADC sampling point. Now we are trying to re-create the original un-sampled waveform - re-creating all the missing bits of data from one sample to the next one. Now the theory is very straightforward - if you use an infinite tap length FIR filter with a sinc impulse response you will absolutely and perfectly reconstruct the bandwidth limited signal - if its perfectly bandwidth limited to below 22.05 kHz it will not matter if you sample at 22 uS or 22 femtoS it will make no difference to the output - if you use an infinite tap length FIR filter. Now of course, we can't have infinite tap lengths filters, we have to make do with something very limited.
 
The question is, what level of time domain accuracy do we need where improving it makes no difference to the sound quality? That's where lots of careful listening tests comes in, as nobody knows. And its where I have been spending a lot of time over the last 18 months working on project xxxx - and I have learnt a lot (and I still have more things to discover, I am sure that I have not gotten to the bottom of the time domain accuracy barrel). What is clear to me, is that the ear/brain is amazingly sensitive to tiny time domain errors - there does not seem to be a level which one can say is insignificant. This is one of the really weird and interesting things about correlating what one hears with real signal errors - the other really odd issue being the perception of sound-stage depth - this can be upset by seemingly impossibly small errors.
 
This is where I find the "DAC bit perfect" concept  - like a cheap politicians sound byte - ridiculous. The job of a DAC is to reproduce the continuous waveform at the ADC sampler - NOT to bit perfectly reproduce the sampled data with all the sampling time domain errors perfectly intact.  
  
If you were to use this filter with a conventional resistor ladder DAC would it work? 
The answer to this is yes, but not as well as Pulse Array - the 16e DAC can reproduce 50 MHz sine wave albeit with 3% THD and noise! The problem with R2R is that the OP can't switch fast enough, as there are a lot of switches involved in the R2R ladder, so in practice you can't run them above 16 FS - but I can run mine at 2048 FS so the digital domain is much closer to the original un-sampled analogue waveform. There are lots of other problems with R2R - noise floor modulation, code dependent glitch energy, high distortion at small signal levels, and moderate distortion at large signal levels.
 
 
I hope I have not confused things too much - but we are dealing with a very complex subject, and something which, after more than 30 years of intense work, I am still learning new things. Things are very complex when you dive into it, and the ear/brain is a remarkably sophisticated device - the illusion of listening to real sounds is a truly amazing brain construct, and its something we know very little about. But at the end of the day, the engineering that goes into Hugo does not matter, its the musicality that counts, so keep on enjoying music! 
 
Rob

 
Quote:
  The earlier prototypes had less current than Hugo - 0.2A RMS. But last Feb we were with Nelson (Malaysian distributor) and I showed him the prototype. He brought some headphones, and we compared it to Hugo - and it was bad, a lot of clipping from Mojo, none from Hugo. It so happened (luckily) Nelson gave us possibly the only 8 ohm headphones on the market Final Audio Pandora.
 
So I upgraded the current to match Hugo, its the same 0.5A RMS. To do this I used 6 small OP transistors in parallel as size would not permit use of the large devices I use in Hugo.
 
Mojo development had some weird fortuitous events - the first time we showed the prototype happened to use the only 8 ohm headphones that would show problems - and the day we were deciding on the design Xilinx emailed me with a new FPGA that would enable Hugo performance but with the needed power, and happened to be in production just in time for Mojo. 
 
Quote:
  I have just measured a Mojo into a 16 ohm load using an APX555 test equipment. With 1% THD 1 kHz single channel,  Mojo delivered 3.30 v RMS - that's 680 mW. Using 50 Hz, it was 668 mW RMS.
 
Rob

  Into 300 ohms, fully charged battery, its 94 mW or 5.3v RMS at the 1% THD point.
 
Rob

  1. The battery is capable of delivering 3A of current, and has very low impedance.
2. Mojo amplifier has a very high power supply rejection ratio.
3. The output is pure class A at 5v RMS into 300 ohm.
4. Reducing the output load only starts to increase distortion with 33 ohms - at this level it is very much lower than other headphone amps. The HD25 is a very easy 70 ohms.
5. Mojo is designed to drive loudspeakers. You will be amazed hearing it fill the room with beautiful sound using efficient 8 ohm horn loudspeakers.
 
Some people like the sound of more distortion - 2nd harmonic fattens the sound making everything sound phat, soft and rounded. But its not natural, nor do I find it musical, as everything sounds phat. I want soft sounds to sound soft, and sharp sounds to sound sharp - not everything to have a soft sheen on things all the time.
 
I can give you another example. I just had an email today from a very experienced dealer that asked me this question:
 
"Chord Mojo should have single amplification which drive to both headphone output, but if I'm using any headphones (HD800 for example, very heavy to drive) to put on headphone output 1 and I connect another headphone to headphone output 2 (Beyer T1 for example), I hear no differences on sound quality. Normally if one amplification section used to drive 2 headphones output, then once we connect the second headphone will make overall sound quality degradation (as the case of Beyerdynamic A20 amp, Grace Design M903, etc). What is the logical explanation of this? As it seems Mojo has 2 separate amplification sections which drive independently for each headphone output."
 
Of course Mojo does not have two amplification stages. It can drive two headphones with ease because it has exceptional low output impedance, and it has exceptional current linearity. So loading it with more headphones has no effect, unlike other headphone amplifiers.
 
Indeed, when I initially started designing Hugo I was shocked how poor from a measurement point of view headphone amps were. Poor output impedance, huge levels of distortion, and poor current linearity seem to be typical. And these things matter, if your goal is transparency and musicality. 
 
Rob

Charging state of the battery makes little difference to the output level

 
 

 
Regarding MFI certification (Mojo)
 

IMPORTANT!: Please be careful with the iOS public betas - they can cause crackling sounds when using Mojo with the CCK
   
(NB: please also view the VIDEOS section!)
 
IMPORTANT : It is YOUR responsibility to make sure that the cable you are buying correctly fits the connectors on your equipment. None of these links are official endorsements.
 

 
Mojo is not Apple MFI-certified (it doesn't have an Apple CCK/MFI chip integrated inside), so connecting Mojo to an iDevice requires a cable with an included MFI chip. This is generally a CCK Lightning to USB Camera Adapter (which must be linked to the male-USB-to-male-microUSB cable that's included with Mojo):

 
I believe a requirement of using the camera kit Apple chip inside your product is complete design disclosure to Apple engineering hardware software the lot. We might be a bit mad but we're not totally crazy. If we had not much technology to hide. Say if we were just using a industry Dac chip we wouldn't have a problem in doing this, but for us today it's a very different story. This is why we will soon offer a plug in module that swallows the official Apple Camera adaptor leaving just the Lightning tail to plug into the I phone.

Apple CCK is a must unless you have a specialty cable with the Apple MFI chip inside.

Your connection should be this:
 
   
.... There is no Apple Co-processor in the Hugo or Mojo so an Apple CCK connector is needed. We (Moon Audio) are MFI certified to build Apple Lightning Cables but Apple will not let anyone build an all in one Lightning CCK cable. trust me we have tried and asked for all kinds of variances on this. This is how Apple makes money on Licensing Co-processor to Dac manufacturers. If we introduce a cable that solves this that reduces the bank role of Apple. Chord will have several accessories down the road the plug into the end of the Mojo, one of which will swallow the end of a CCK. See my Mojo review here of pics: http://blog.moon-audio.com/chord-mojo-review/

 
 
Moon-Audio's video overview of connecting Mojo with Apple's CCK cable
 
 
If you dislike using the Apple CCK connector, there are some 3rd-party cables which circumvent this, and thereby allow you to use a single cable. However, they are not Apple-certified, so there is a small risk that Apple may find a way to stop them working, in a future iOS update. At this point in time, though (iOS 9.3), many people are successfully using them. Ultimately, please do your own research before buying/trying any of these CCK-circumvention cables!!
 
Lavricable (circumvents Apple CCK)
 
CAUTION!  This cable does not appear to function well with iOS 10, so probably best to choose an alternative until (IF) the issue is resolved

 
lavricables@gmail.com
www.ebay.com/itm/Pure-Solid-Silver-Lightning-to-Chord-Hugo-Mojo-interconnect-cable-Iphone-4-5-6-/172223242191
 
 
ZY Cable Lightining to MicroUSB cable (circumvents Apple CCK)
 
WARNING! - Although this (seemingly-generic) cable has worked for many people, reports have begun to come in, during Jan 2017, that it may now not be working
 
http://www.head-fi.org/t/784602/chord-mojo-dac-amp-faq-in-3rd-post/28275#post_13142121
http://www.head-fi.org/t/784602/chord-mojo-dac-amp-faq-in-3rd-post/28530#post_13150044
 
This looks to be identical to the Hi-FiSpot cable, listed beneath this one
 
CAUTION!  The thin braided wire used in this cable may not be the most resilient to wear&tear
 
Quote:
Just to close the loop, I received ZY Cables' Lightning to MicroUSB cable yesterday (along w/ AQ Nighthawks! :)). Purchased here. So far, no issues on iPhone 7+ 128GB running iOS 10.1.1. Sample rate reflected correctly on Mojo when playing high-res from Onkyo HF, NePlayer and HibyMusic. Seems to work well (have been running in Airplane mode so don't know about RF rejection). Will see what happens with the next iOS update... Thanks all.
 

 

 

Available from:
www.ebay.com/itm/112101143986
 
Hi-Fi Spot Lightning Cable
 
WARNING! - Although this (seemingly-generic) cable has worked for many people, reports have begun to come in, during Jan 2017, that it may now not be working
 
http://www.head-fi.org/t/784602/chord-mojo-dac-amp-faq-in-3rd-post/28275#post_13142121
http://www.head-fi.org/t/784602/chord-mojo-dac-amp-faq-in-3rd-post/28530#post_13150044
 
This looks to be identical to the ZY cable, listed above
 
CAUTION!  The thin braided wire used in this cable may not be the most resilient to wear&tear

www.ebay.com/itm/Lightning-Line-Out-Dock-to-MICRO-USB-cable-for-hugo-mojo-AMD-n5-iphone-5-5S-6-6S/321954079094
 
Taobao iPhone654-HugoPHA-mojo-lightning-kit otg ios9 (circumvents Apple CCK)

Above image credit: AudioBear
 

world.taobao.com/item/44240667193.htm
 
Fiio L19 (circumvents Apple CCK)
 
Note: some people feel this cable may not be the most reliable choice
 

penonaudio.com/L19-Lightning-to-Micro-USB
 
 
Quote:
The Fiio L19 isn't a Fiio product, I've had this confirmed by Fiio direct. Although I don't believe there are any issues with the cable.

 
Video of Fiio L19 cable working with iOS 9.2 and Mojo
 
 
 
Penon Audio  Lightning to Micro USB Hugo/Mojo/PHA DAC Audiophile Pure Silver Decoding Cable (circumvents Apple CCK)
 
This cable seems to be a popular & reliable choice, BUT PLEASE SPECIFY RIGHT-ANGLED PLUGS, to reduce strain on Mojo & iDevice sockets


 
Right-Angled version:

 
Available from:
 
http://penonaudio.com/Lightning-Pure-Silver-Decoding-Cable
 
 
Zee's Music braided with 8 cores OCC + gold plated connectors & WBT 4% silver solder (circumvents Apple CCK)

Available from:
www.ebay.com/itm/252476440532
 
 
 
We at chord looked at those adaptors but decided they were very likely to potentially put a lot of sheer force right into the most fragile part of to USB Connectors in our unit. this is why we made our adaptor substantially larger to accommodate tour points of connection into Mojo and a cupped end to ensure damaging forces are less likely to be transferred into any single USB socket.

 

 
 
 
  1. Also of interest: www.head-fi.org/t/784602/chord-mojo-the-official-thread-please-read-the-3rd-post/10860#post_12323045
 
 
Please note: Chord Electronics does not specifically endorse any cables which seek to circumvent the CCK, so cannot be held responsible for any issues arising therefrom.
 
 

 
  1. If you have problems with intermittent connection, read this and please note that even genuine Apple CCK lightning cables can sometimes be faulty
 
 
 

Mojo owners using iOS will need to use a software app, in order to output Hi-Res audio through the Lightning connection. There are a few options:
 
  1. Onkyo HF Player ($10 for HD Version)
 
  1. HiBy Music (free)
 

 
  1. Physical attachment without obscuring screen: www.head-fi.org/t/784602/chord-mojo-the-official-thread/2940#post_12033020
 

 

(NB: please also view the VIDEOS section!)
 
IMPORTANT : It is YOUR responsibility to make sure that the cable you are buying correctly fits the connectors on your equipment. None of these links are official endorsements.
 
 
http://www.amazon.com/gp/product/B00FF086HE
 
Sony WMC-NWH10 USB Conversion Cable for Hi Res Audio Output
 

Image credit: Whitigir
 
 
Those who need a cable for digital from zx2/Sony Walkman and can deal with Chinese:

http://m.intl.taobao.com/detail/detail.html?spm=a1z5f.7632060.0.0&id=45034655500

I got one and been using it
biggrin.gif
. Cheap too

 
 
 
  Sony Walkman to HUGO/ Mojo usb connection cable alternative.
 
.... just wanted to share some info which some may find useful if using a Sony Walkman with the Hugo as if like me you have got annoyed at using the special Sony WMC-NWH10 USB conversion cable where you then have to still use a normal usb cable into this bulky not best quality adaptor Sony have ever done which makes it very messy and cumbersome and untidy when strapping the Sony ZX1/ ZX2 to the Hugo. 
  I had already tried a aftermarket cable by Music heaven .... Good news is he has managed to get the Sony NWH10 plug to work with a cable with micro usb on the end to work with his Sony ZX2 and ALO successfully and is willing to make some up for anyone who needs one.  
 
Anyone who is interested just PM Wfanning1 and have a chat with him. 
 
Here are some pics he sent me only the other day of the cable which he is classing as prototypes as he said he will be able to put different cables on with either straight or single ended usb: 
 

 

 

 
 
 
 
 
Also see:
 
www.head-fi.org/t/784602/chord-mojo-the-official-thread-please-read-the-3rd-post/10845#post_12322147
 
 
 
  This cable looks really interesting also!
 
http://www.amazon.com/Custom-Walkman-Digital-WM-Port-Degrees/dp/B00YWEHSQY/ref=sr_1_2?ie=UTF8&qid=1445462623&sr=8-2&keywords=WMC-NWH10

This will not work with HUGO/Mojo. It works only with OPPO HA-2. There's a 20K Ohm resistor between pins 4 and 5.
  .... I did try that cable from Amazon. For this cable to work with Oppo, they have included a 20K Ohm resistor between pins-4 and 5 on the micro-usb side.
I remove the resistor and shorted pins-4 and 5, to see if it will work with HUGO. It didn't.
 

   
 
 
 
We at chord looked at those adaptors but decided they were very likely to potentially put a lot of sheer force right into the most fragile part of to USB Connectors in our unit. this is why we made our adaptor substantially larger to accommodate tour points of connection into Mojo and a cupped end to ensure damaging forces are less likely to be transferred into any single USB socket.

 

 
 
 

(NB: please also view the VIDEOS section!)
 
IMPORTANT-1 : It is YOUR responsibility to make sure that the cable you are buying correctly fits the connectors on your equipment. None of these links are official endorsements.
 
 
 
IMPORTANT-2:  PLEASE CHECK that your Android smartphone supports USB OTG output!! Not all Android phones have this functionality properly implemented. A good place to start, would be here:
 
www.extreamsd.com/index.php/2015-07-22-12-01-14/usb-audio-driver
www.head-fi.org/t/595071/android-phones-and-usb-dacs
 
 
 
IMPORTANT-3:  Please note that you need an OTG cable, not just a standard microUSB cable (although they may look identical, the pin wiring differs):
 
  A USB OTG cable (with a "ID pin 4-connected to-pin 5" micro USB plug) is needed to connect an Android device to a standard USB DAC.
The dual-role Android device is configured to USB host mode (able to interwork with a USB peripheral like Mojo) on the insertion of a USB OTG plug
 
More details can be found at:
www.head-fi.org/t/595071/android-phones-and-usb-dacs
Android USB audio
. FAQ
. A list of USB OTG cables / adapters

 
 
Economical 7cm-long Male-MicroUSB-B-to-Male-MicroUSB-B OTG cable (link)

Please note: although these cheap generic Chinese cables can work OK, the connector quality may sometimes cause an unreliable connection!
 
 
Economical 10cm-long Male-MicroUSB-B-to-Male-MicroUSB-B OTG cable
Available from eBay or aliexpress

Please note: although these cheap generic Chinese cables can work OK, the connector quality may sometimes cause an unreliable connection!

 
ToddTheVinylJunkie (TTVJ) short Male-MicroUSB-B-to-Male-MicroUSB-B OTG cable (Link)

 

 
PenonAudio MICRO USBOTG Silver (Link)


Image credit: sonickarma
 
NOTE!: PenonAudio now offer a Right-Angled plug option, which will be less stresseful for Mojos input socket

 
Available from: http://penonaudio.com/OTG-Pure-Silver-Cable
 
 
 
 
 
 
 
 
                     (image credit: 'Hawaibadboy')
 
 
Moon-Audio (USA) offer premium 7.5cm-long Male-MicroUSB-B-to-Male-MicroUSB-B OTG cable (Link)
 
Custom-Cable (UK) offer premium 25cm-long Male-MicroUSB-B-to-Male-MicroUSB-B OTG cable (Link)


 
Do you need Type-C to microUSB?
 
Monoprice Select Series 2.0 USB-C to Micro B Cable, 6-inch

www.monoprice.com/mobile/product/details/13013
 
 

https://item.taobao.com/item.htm?spm=a1z09.2.0.0.qO1xhm&id=537999317212&_u=s11o9rqq6d35 
 
 

https://item.taobao.com/item.htm?spm=a1z09.2.0.0.JuZF4I&id=529538146962&_u=s11o9rqq4e5b
 
 
 
NOTE: Always be sure that you are using your OTG cable the correct way around! (OTG plug at the phone end, not at the Mojo end)
 
 

 
 
Mojo owners using Android smartphones will need to use a software app to bypass Androids automatic 24/192 upsampling (please also see the 'Informative posts by Rob Watts' section, higher-up this post).
 
 
 

Quote:
  If the native USB audio does not work then you need to use one of the third-party USB audio music players (which include their own USB audio function / soft driver).
 
More details can be found at:
http://www.head-fi.org/t/595071/android-phones-and-usb-dacs

 
 
 
 
There are several apps currently available:
 
  1. USB Audio Player Pro (UAPP) (the most popular - approx. $8) (compatibility-list, & very useful overview, on UAPPs homepage)
 
  1. Onkyo HF Player (approx. $8 for Hi-Res version)
 
  1. HiBy Music (free)
 
 
 
 
 
1) Enter UAPP:
 

 
 
2) Select 'Artists' page:
 

 
 
3) Select drop-down menu:
 

 
 
4) Select Tidal:
 


 
 
You should see this screen:
 

...but if the password screen doesn't appear, then just click on the 'human' icon like this:

 

 
Also, remember to set the quality level:
 



Raw screengrabs credit: maxh22
 
 
 
Note 3 has this weird usb connection.So if I get a Note 3 I will have to get a special usb connection to connect to the Mojo ?


 
that is usb 3.0, you can use standard micro usb 2.0 while pairing note 3 with mojo

 

 
 
  I connected my Mojo to my Samsung S7 and opened USB player audio pro...i get the error 'error initalizing usb system'

 
 
Quote:
  fixed it:
 
ran this
 
https://play.google.com/store/apps/details?id=org.tauruslabs.usbhostcheck

 
 
 
  1. Problems getting Samsung Note 4 to play with Mojo?
 
  1. Problems getting Samsung GS3 LTE to play with Mojo?
 
  1. Information on Android ROMs, in relation to USB Audio
 
 
 
 
We at chord looked at those adaptors but decided they were very likely to potentially put a lot of sheer force right into the most fragile part of to USB Connectors in our unit. this is why we made our adaptor substantially larger to accommodate tour points of connection into Mojo and a cupped end to ensure damaging forces are less likely to be transferred into any single USB socket.


 
 
 
 
 
Here's a wildcard some of you may care to look into: www.head-fi.org/t/784602/chord-mojo-the-official-thread-please-read-the-3rd-post/13440#post_12425559
 
NB: Mojo USB input is:
 
  USB is isochronous asynchronous. This means that the FPGA supplies the timing to the source, and incoming USB data is re clocked from the low jitter master clock. So again source jitter is eliminated.
 
Rob

 
 
  The problem with galvanic isolation is that the USB decoder chip and the data transmitter is powered from the mobile phone. There is no point in having 8 hours on Mojo if your mobile phone battery is depleted earlier. That said, you really need isolation when using a PC; its an order of magnitude smaller problem when using mobile phones, as everything is battery with no ground loops, and mobiles are incredibly power efficient compared to PC's - which gives you very much lower RF noise.

Quote:
  To eliminate the RF & signal correlated noise on USB you need galvanic isolation. The downside to galvanic isolation is that it draws power from the source - which is not something we can do with a mobile product. All Chord desktop DAC's have USB galvanic isolation now.
 
That said, mobile sources are much lower noise - they have very efficient processors, unlike a PC, and there is no ground, so circulating currents are much less, so it is a much smaller problem with mobile. If you can do it, use the optical, as this usually sounds the best and is completely isolated. Optical has a undeservedly poor reputation, as it sounds much smoother and darker than other inputs, and this is just a feature of lower noise floor modulation - its smoother with better instrument separation and focus - but lack of glare is often confused with a lack of detail resolution. Listening tests must be done with a lot of care, as it is easy to draw the wrong conclusions!
 
Rob

 
  .... Mojo has the same USB as Dave - but with Dave it is galvanically isolated, so RF noise and signal correlated currents from the source can't upset Dave at all. I can't do this with Mojo as it draws too much power from the device connected to the USB - but the upside is that mobile sources create much less noise as they are power efficient and there is no ground loops either due to battery operation.
 
So Mojo like Dave has solved the jitter problem via USB, as the timing for the data comes from Mojo not the source.
 
Rob

 
 

(NB: please also view the videos section!)
 
IMPORTANT : It is YOUR responsibility to make sure that the cable you are buying correctly fits the connectors on your equipment. None of these links are official endorsements.
 
 
www.head-fi.org/t/784602/chord-mojo-the-official-thread/2940#post_12033270
 
www.head-fi.org/t/784602/chord-mojo-the-official-thread/2595#post_12029029
 
www.head-fi.org/t/784602/chord-mojo-the-official-thread-please-read-the-3rd-post/12810#post_12407306
 
www.head-fi.org/t/784602/chord-mojo-the-official-thread/3570#post_12046140
 
www.head-fi.org/t/784602/chord-mojo-the-official-thread-read-the-first-post-for-updated-info/4020#post_12054119
 
www.head-fi.org/t/784602/chord-mojo-the-official-thread-read-the-first-post-for-updated-info/4275#post_12057847
 
Please be cautious about head-fier derGabe selling cables  - some members have not received their orders!
 
www.head-fi.org/t/784602/chord-mojo-the-official-thread-please-read-the-3rd-post/11490#post_12349732
 
 
 
iBasso DX90 issues:  www.head-fi.org/t/784602/chord-mojo-the-official-thread-read-the-first-post-for-updated-info/4800#post_12069924
 
 

 
Fiio x3ii and X5ii  owners, please additionally note: www.head-fi.org/t/784602/chord-mojo-the-official-thread-please-read-the-3rd-post/14070#post_12442957
MoonAudio offer suitable cables
Custom-Cable also offer suitable cables

 
[spoiler='Uranus' Co-Axial cable for Fiio X3ii and X5ii] 'Uranus' Co-Axial cable for Fiio X3ii and X5ii

(image credit: 'noobandroid')
 
NB: the DAC end of noobandroid's cable (RH, in the pic) has one too many poles, but, if wired appropriately, may still work correctly (please see the 2nd connection image immediately below). tkteo's Dyson version is what one would actually expect.
 
 
Here are the signal paths for an appropriate Co-Axial cable to connect Fiio X3ii and X5ii DAPs to Mojos Co-Axial digital input:
 

(pin-out identities based upon these: www.head-fi.org/t/784602/14985#post_12467535 )
 
IMPORTANT: James (CEO of Fiio) has privately confirmed to me that the above diagram is definitely correct for X3ii, X5ii, and X7
 
 
Alternatively, if you wish to use a stereo TRS plug at the Mojo end (instead of the mono TS plug in the above picture, which is really all that is required), then the pin connections would be as follows:
 

(pin-out identities based upon this: www.head-fi.org/t/784602/18675#post_12651727)
 
 
  Lately, I have been recommending a certain type of TRS Connector(or Plug) for the coaxial cable to connect a Fiio X3ii/X5ii/X7 to a Mojo, and it appears to be causing some confusion here. Hopefully this post will help clarify this confusion.
 
PLEASE NOTE: This post only applies to Fiio X3 2nd Gen, X5 2nd Gen and X7 devices. Before we get into the topic, here is a picture to understand what TS, TRS, TRRS Connectors are:
 
 
Fiio X3ii, X5ii and X7 devices use a single 3.5mm port that is shared for both Line-Out and Coaxial Digital-Out purposes. A regular coax cable will NOT work with these devices. You need a modified cable with 3.5mm TRRS Connector to work with these devices. The TRRS pins on this coaxial cable have to be configured as shown in the picture below:
 
 ​
 
Now, the coax port on the Mojo is a simple, dedicated 3.5mm coax port. So all it needs is a 3.5mm TS Connector. So in order to connect a Fiio X3ii/X5ii/X7 to a Mojo, you need a coax cable with 3.5mm TRRS Connector on one end and, a 3.5mm TS Connector on the other. The TRRS end goes into the Fiio devices and the TS end goes into the Mojo.
 
EDITOR'S note: There are cable makers who sell this modified cable in both Straight Connectors and Right-Angle Connectors version, but it can be necessary to make a special-order. Dyson Audio used to make them, but this maker became unreliable, and is best avoided until further notice.
 ​
 
If you want a cable with Straight Connectors on both end, you can stop right here and can right away purchase it from one of these places:
Moon-Audio (US) / Custom-Cable (UK) / Uranus (Malaysia)
 
Now the PROBLEM arises, when you want this cable with Right-Angle Connectors. For some reason, the only Right-Angle version of TS Connector, that seems to be available in the market is this plastic one:
 
 
I did not like this plastic connector. I looked around for a solution and I found out that, cable makers like Moon-Audio and Uranus, used a TRS Connector instead on the TS Connector, in their Right-Angle version of the coax cable. But now I had ANOTHER PROBLEM. Uranus-Cable does not ship outside Malaysia and the cable from Moon-Audio was too expensive.
 
  Hope this helps!
 
-EagleWings

 
 
 

(NB: please also view the videos section!)
 
IMPORTANT : It is YOUR responsibility to make sure that the cable you are buying correctly fits the connectors on your equipment. None of these links are official endorsements.
 
  1. Mojos optical input is a standard Toslink Optical socket. However, many DAPs have optical outputs using 3.5mm sockets, so PLEASE CHECK before buying an optical cable
 

 
 
Sysconcept custom-made low-profile optical cable
 
www.head-fi.org/t/784602/chord-mojo-the-official-thread/2370#post_12025183
 
http://www.sysconcept.ca/product_info.php?products_id=349
 

 
 

 
 
Google Chromecast Audio optical cable (6 inches)
 
https://store.google.com/product/optical_cable_chromecast_audio
 

 
 
 
 
 
www.head-fi.org/t/784602/chord-mojo-the-official-thread/2880#post_12032024
 
www.head-fi.org/t/784602/chord-mojo-the-official-thread/2265#post_12023214  (specifically to a Mac computer)
 
related (non-Mojo-specific) discussion about optical cables: www.head-fi.org/t/784602/chord-mojo-the-official-thread/2250#post_12023147
 
[/spoiler]
 

 
www.head-fi.org/t/784602/chord-mojo-the-official-thread/2025#post_12020136
 
www.head-fi.org/t/784602/chord-mojo-the-official-thread-read-the-first-post-for-updated-info/4770#post_12069520
 
www.head-fi.org/t/784602/chord-mojo-the-official-thread-please-read-the-3rd-post/12360#post_12394268
 
www.head-fi.org/t/784602/chord-mojo-the-official-thread-please-read-the-3rd-post/12855#post_12408223
 
....  
And BTW to clear up the Windows Phone confusions, it's confirmed otg is only supported on newer models that supports 'Continuum' for example the Lumia 950/xl. I am not sure about older models but my Lumia 830(running the latest windows 10m preview) does not work with the mojo.

http://forums.windowscentral.com/microsoft-lumia-950/395360-external-audio-dac-support-usb-otg.html

 
 

 
Blackberry Phone
 

 
www.head-fi.org/t/784602/chord-mojo-the-official-thread-read-the-first-post-for-updated-info/5070#post_12078284
 
 
Almost any DAP with a digital-output should function correctly with Mojo.
Some that Head-fiers have successfully used with Mojo include:
 
Fiio X3ii
Fio X5ii
Fiio X7
AK70
AK100
AK120
AK240
AK320
AK380
Shanling M1
Soundaware M1 Esther
 
Also see: www.head-fi.org/t/784602/chord-mojo-dac-amp-faq-in-3rd-post/19620#post_12706423
 

 
 

 
 
ALSO SEE (regardless of which playback software is used): www.head-fi.org/t/784602/chord-mojo-the-official-thread-please-read-the-3rd-post/9780#post_12277215
 

 
 
 
 
  1. Silicone wristbands
  2. Large O-rings from an automotive spares supplier (or these, although they may not last very long)
  3. Sticky back velcro (may leave sticky residue on devices, in the event of trying to remove it entirely)
  4. 3M Dual Lock Low Profile
  5. Double-sided self-adhesive foam (may leave sticky residue on devices, in the event of wishing to remove it entirely) If you use single-sided, you'll still need silicone bands as well
  6. Blu-Tack
  7. 3M 'Command' picture-hanging strips
  8. Cured silicone (just like single-sided foam, this will need silicone bands, as well)
  9. Thin bead of silicone sealant (warning: don't use this method unless you are familiar with how silicone can be removed correctly)
 
 
 
 
or an aluminium case
 
 

 
 
 
 ​
 ​
Chord Mojo + AK70 Music Player Leather Case by Miter
$99.00
 ​
Free U.S. Ground Shipping
 ​
 
 
 
 
 
 
 ​
 ​
Free U.S. Ground Shipping
 ​
 ​
$99.00 Chord Mojo + AK70 Music Player Leather Case by Miter
 
The Chord Mojo + AK70 Music Player Leather Case by Miter made in South Korea, for protecting your Astell & Kern AK70 and Chord Mojo from scratches, shocks and fingerprints.

  1. Hand Crafted MITER Leather Case for Chord Mojo + Astell & Kern AK70 Music Player
  2. Patented / Steel Frame Stand & Case Cover
  3. Protects from Shocks, Fingerprints and Scratches
  4. Hand crafted with Superior craftsmanship
  5. Made in South Korea
  6. Made with “Oil Pull-up” leather, a favorite of Miter, due to its beautiful matte finish, unique in its characteristics with respect to the oily finish different from other varieties of leathers.

 

 

Thank You.​
Drew Baird
Moon Audio 
106 Brady Court
Cary, NC 27511 
919-649-5018 
Drew@moon-audio.com
 
HeadFiLogoSignature.jpg
Follow us Today!​
     

 
 
 
 
Chord have released a high-quality case of their own (please see 'Official Chord Accessories for Mojo', in next section)
 
   
 
Dignis has come up with some cases for the AK70/Mojo combo which looks rather nice.

Photos taken from Dignis Japan's Twitter page.




 
 
 
 
 
Here is a way to seal unused ports on Mojo
 

 
 

 
 



 
 
More pics: www.head-fi.org/t/784602/chord-mojo-the-official-thread-please-read-the-3rd-post/14715#post_12459171
 
Chord case review: https://www.youtube.com/watch?v=J_VuZJH5yvU
 
 
 
 


 
The BASIC USB Adapter Module, that is included in the official Cable Accessory Pack (as shown in the above contents), is used as shown in the following image.
 
It houses the connector plug of the digital transmission cable and thereby reduces stress upon that connector plug and Mojos connector socket.
 
When attached to Mojo, the module also makes it easier to stack with a smartphone without obscuring the screen, since the combination of Mojo+module more closely matches the length of a smartphone.

 
 
blog.moon-audio.com/chord-mojo-review/
 




images credit: http://blog.moon-audio.com/chord-mojo-review/
 


www.modernhifi.de/chord-mojo-zubehoer/  


 
images credit: www.modernhifi.de/chord-mojo-zubehoer/
 
 
 
NB: please also see the Munkonggadgets interview with John Franks
 
 
 
 
 
 
  @Mojo ideas Is there any chance of you making a case that includes the Mojo with the add on module?

 
Mojo Ideas replied to me when I asked if there would be a case to encompass Mojo and SD-card module. Answer, "Yes".

 
 
There was a bit of non-official discussion on this, a year ago - Chord may have different plans now (perhaps not so many modules on the drawing board), but I still anticipate an extended case of some description, to see the light of day, in due course:
 
 
 
. This one is option 3 our favourite option one was disregarded earlier


My vote also goes to this design.

I'd like the silver Mojo which seems to be lurking inside as well please.:hugging:

However what happens (to either solution) if someone wants to clip on one of the proposed add-on modules?

Ian

 
 
I don't anticipate that those cases will be particularly expensive - therefore, it might be quite viable to produce standard-sized versions and also extended versions, to simultaneously encompass both the Mojo and an attached add-on module. This could also confer a nice benefit of reinforcing the 'join' between the Mojo and the add-on.

 
 
 

 


 





(Image credit - Drew  @ Moonaudio)





This is just for fun, so excuse the roughness of this photoshopped mock-up, but this is the kind of thing I was imagining:


Already imagined and more but it can't be quite like that as there are four different module to accommodate it's being refined and costed but it's great you guys can get a hint of where we are going with mojo it's more of a system that a product. But please be patient we want to get all this really right and not just okay.

 
 
 
 
 
The following comment, from John Franks, is mid 2016:
 
my thoughts, too....esp for $100

and if that add-on comes out shortly
what happens then?


Because we have the extender module and the more complex modules underway it's was always our plan to offer a well made longer case too.

 

They are currently in manufacture and you've reminded me to chase the guys.

 
 
 
 

 
This will always be a contentious issue (such is the nature of geeky audiophiles
wink_face.gif
). There will always be fans of one connection type or another.
 
Here are some aspects of the discussion:
 
Rob has a preference for optical on Hugo, but USB on Hugo TT. The difference between them is that TT has galvanic isolation.

Mojo and Hugo don't use galvanic isolation as this would suck power out of the mobile device that they are connected to. You would drain the battery in your phone more rapidly if it had galvanic isolation.

TT, being for home use, doesn't care if it drains power from the device that's connected to its "USB HD" input.

The "USB SD" input on TT doesn't have galvanic isolation and is there for phones. It also works on Windows PCs that do not have the Chord driver installed.

 
  The problem with galvanic isolation is that the USB decoder chip and the data transmitter is powered from the mobile phone. There is no point in having 8 hours on Mojo if your mobile phone battery is depleted earlier. That said, you really need isolation when using a PC; its an order of magnitude smaller problem when using mobile phones, as everything is battery with no ground loops, and mobiles are incredibly power efficient compared to PC's - which gives you very much lower RF noise.

 
Quote:
  To eliminate the RF and signal correlated noise on USB you need galvanic isolation. The downside to galvanic isolation is that it draws power from the source - which is not something we can do with a mobile product. All Chord desktop DAC's have USB galvanic isolation now.
 
That said, mobile sources are much lower noise - they have very efficient processors, unlike a PC, and there is no ground, so circulating currents are much less, so it is a much smaller problem with mobile. If you can do it, use the optical, as this usually sounds the best and is completely isolated. Optical has a undeservedly poor reputation, as it sounds much smoother and darker than other inputs, and this is just a feature of lower noise floor modulation - its smoother with better instrument separation and focus - but lack of glare is often confused with a lack of detail resolution. Listening tests must be done with a lot of care, as it is easy to draw the wrong conclusions!
 
Rob

 
 
 
  USB DAC data is not bit perfect by any stretch of imagination. It tries to be bit perfect, but if it fails to be, it is not corrected at any point. Though again, there should not be any ground and sky differences. 

That is not the case with Chord's windows drivers. If faulty data is sent through, then the DAC requests a repeat, and so ensures perfect data transfer.
 
It is possible with all other OS; but having said that, the data failure rate is very low (otherwise DoP would not work).
 
The USB connection making a difference to the sound is not data related - its down to RF and correlated noise (not jitter as this is completely removed too) - take a look at my previous posts if you are interested. 
 
Rob

 
 
I don't think anyone will ever answer that question satisfactorily if you know how the technical aspects of the devices work. The transport is passing a purely digital signal onto the Mojo, just what the original file is. Mojo does all the work. That being said, some people swear they hear a difference in sound depending on the transport. You decide what is most likely:)

The reasons why sources and digital interconnects sound different are well understood - see some of my posts. In a nutshell it is not jitter (all my DACs are source jitter intolerant) but down to RF noise and distorted currents from the source flowing into the DAC's ground plane. The RF noise inter-modulates with the analogue electronics, creating random noise as a by product, which creates noise floor modulation, and that makes it sound brighter or harder. The correlated or distorted currents very subtly add or subtract to small signals, thus changing the fundamental linearity, which in turn mucks up depth perception.
 
But I also agree in that lots of people hear changes that are not there - I for one have never heard any difference with optical cables (assuming all are bit perfect) with my DAC's, but lots of folks claim big differences. Placebo, or listening with your wallet, plays a part here. Then there are cases of people preferring more distortion... Listening tests must be done in a very controlled and careful fashion, particularly if you are trying to design and develop things.
 
Rob

 
 
 
  Digital transmission is based on SPDIF standard which transmits data and clock information as an encoded signal usually using PCM, that information is decoded on the Mojo into data and clock signal so it's important that the encoded information be jittered free and not degraded over short distance.
 
The USB transmission on the other end is a device to device transmission mechanism using an encoding scheme and handshaking mechanism, it is usually stream based so more tolerant to poorer wire as frames are transmitted and decoded from the source to the target device. The target device will reconstruct the data and clock signal from the frame and then feed it to the DAC to be analog reconstructed and eventually band pass filtered to remove any residual high and low frequency signals out of the audio band.I still think you need to keep the USB cable short but it is more tolerant of longer lengths up to a limit.
 
To make a story short, the short USB cable is fine but an analog cable used as a digital one is just a bad idea. Again, that's just my opinion.


Just to clarify:
 
1. SPDIF decoding is all digital within the FPGA. The FPGA uses a digital phase lock loop (DPLL) and a tiny buffer. This re-clocks the data and eliminates the incoming jitter from the source. This system took 6 years to perfect, and means that the sound quality defects from source jitter is eliminated. How do I know that? Measurements - 2 uS of jitter has no affect whatsoever on measurements (and I can resolve noise floor at -180dB with my APX555) and sound quality tests against RAM buffer systems revealed no significant difference. You can (almost) use a piece of damp string and the source jitter will be eliminated.
 
2. USB is isochronous asynchronous. This means that the FPGA supplies the timing to the source, and incoming USB data is re clocked from the low jitter master clock. So again source jitter is eliminated.
 
So does this mean that any digital cable will do?
 
Sadly no. Mojo is a DAC, that means its an analogue component, and all analogue components are sensitive to RF noise and signal correlated in-band noise, so the RF character of the electrical cables can have an influence. What happens is random RF noise gets into the analogue electronics, creating intermodulation distortion with the wanted audio signal. The result of this is noise floor modulation. Now the brain is incredibly sensitive to noise floor modulation, and perceives this has a hardness to the sound - easily confused as better detail resolution as it sounds brighter. Reduce RF noise, and it will sound darker and smoother. The second source is distorted in band noise, and this mixes with the wanted signal (crosstalk source) and subtly alters the levels of small signals - this in turn degrades the perception of sound stage depth. This is another source of error for which the brain is astonishingly sensitive too. The distorted in band noise comes from the DAP, phone or PC internal electronics processing the digital data, with the maximum noise coming as the signal crosses through zero - all digital data going from all zeroes to all ones. Fortunately mobile electronics are power frugal and create less RF and signal correlated noise than PC's. Note that optical connection does not have any of these problems, and is my preferred connection. 
 
Does this mean that high end cables are better? Sadly not necessarily. What one needs is good RF characteristics, and some expensive cables are RF poor. Also note that if it sounds brighter its worse, as noise floor modulation is spicing up the sound (its the MSG of sound). So be careful when listening and if its brighter its superficially more impressive but in the long term musically worse. At the end of the day, its musicality only that counts, not how impressive it sounds.         
 
Rob

 
 
 
Two questions related to Rob Watts' comments on optical output as a source to the Mojo;

1. What is audio "glare"? What does it sound like? And how does one distinguish it from detail?

2. Is the conclusion that one with an Apple computer should be using its optical/toslink output (that also serves as a headphone output) rather than its USB output, even if one uses an AudioQuest Jitterbug, Schiit Wyrd, Uptone Regen and an Akiko USB tuning stick, or similar devices ?

Glare is normally used for extreme form of hardness or grain in the treble. So I guess one could say going from bad to good glare, grainy, hard, bright, smooth, dark. 
 
Distinguishing it from detail is tricky as a brighter sound is easy to confuse it with more detail resolution. Indeed, truly more transparency, does sound brighter. So you have to be very careful, and I have been caught out in the past. One way of recognising it is with timbre - it the extra brightness is noise floor modulation for example, then all instruments will sound brighter - even those that are supposed to sound rich and dark. But if the brightness is better detail resolution, then smooth instruments will just sound clearer, not brighter. Also, if instrument separation and focus is worse, then it is not more transparency.
 
When somebody says it sounds better, but can't actually describe in details what the differences are, be warned! They may be preferring distortion. Fortunately, our lizard brain ignores all this - if its really better, it will be more emotional and involving, so you should use this as your goal. But assessing whether its more emotional or musical takes a lot of time, you can't do it on a quick AB test.
 
The USB filter devices help (hopefully) but do not solve the problem. It has to use galvanic isolation to do it properly.
 
Rob

 
There are pros and cons to each connection type. Optical may be the most immune to RF, but most people encounter no RF issues with USB or Co-ax, so it is best to choose your connection type based upon what is offered by your existing player, and then to learn how to get the most reliable result from that connection type. Mojo itself has the potential to get equally-superb SQ from Optical, Co-ax, or USB.
 
  Just to make it 100% clear - the USB input will measure absolutely identically to the coax or optical inputs if the USB data is bit perfect.
 
I have set up my APX555 so that it uses the USB via ASIO drivers, and I get exactly the same measurements on all inputs - 125 dB DR, THD and noise of 0.00017% 3v 1k 300 ohms. I have done careful jitter analysis, FFT analysis down to Mojo's -175dB noise floor, and can measure no difference whatsoever on all inputs (with the APX always grounded on the coax).
 
If somebody does measure a difference its down to mangled data on the USB interface (or perhaps poor measuring equipment - Mojo is way better than most test equipment). Mojo can't convert 16 bit data back to 24 bit....
 
Rob 

 
 
If you're still wondering about the very small percentage of RF issues, then please skip down 6 sections below
 

 
Android automatically upsamples ALL music files to 24/192, which is not a good thing for Mojo - here's what to do about it (IMPORTANT!: please see discussion in the 'Connecting Mojo to OTG (microUSB) devices' section, above)
 

 
1) Enter UAPP:
 

 
 
2) Select 'Artists' page:
 

 
 
3) Select drop-down menu:
 

 
 
4) Select Tidal:
 


 
 
You should see this screen:
 

...but if the password screen doesn't appear, then just click on the 'human' icon like this:

 

 
Also, remember to set the quality-level:
 



Raw screengrabs credit: maxh22
 

 
www.head-fi.org/t/784602/chord-mojo-the-official-thread/195#post_11994409
 
www.head-fi.org/t/784602/chord-mojo-the-official-thread-read-the-first-post-for-updated-info/4200#post_12056432
 
www.head-fi.org/t/784602/chord-mojo-the-official-thread-read-the-first-post-for-updated-info/4725#post_12068719
 
www.head-fi.org/t/784602/chord-mojo-the-official-thread-please-read-the-3rd-post/8340#post_12217948
 
www.head-fi.org/t/784602/chord-mojo-the-official-thread-please-read-the-3rd-post/9750#post_12274956
 
Thanks to 'DanBa' for this informative post:  www.head-fi.org/t/595071/android-phones-and-usb-dacs/7350#post_12061822
 
 
Also consider:
 
www.head-fi.org/t/784602/chord-mojo-the-official-thread-please-read-the-3rd-post/13725#post_12433418
 
www.head-fi.org/t/784602/chord-mojo-the-official-thread-please-read-the-3rd-post/13800#post_12435426
 
www.head-fi.org/t/784602/chord-mojo-the-official-thread-please-read-the-3rd-post/13575#post_12428810
 
www.head-fi.org/t/784602/chord-mojo-the-official-thread/2280#post_12023830
 
www.head-fi.org/t/784602/chord-mojo-the-official-thread/2910#post_12032759
 
www.head-fi.org/t/784602/chord-mojo-the-official-thread-please-read-the-3rd-post/14775#post_12460426
 
If you're using Neutron & can't output higher than 44.1khz
 
If you are experiencing clicking sounds, you may need to adjust buffer settings:
 
www.head-fi.org/t/784602/chord-mojo-the-official-thread-please-read-the-3rd-post/9930#post_12282635
 
www.head-fi.org/t/784602/chord-mojo-the-official-thread-please-read-the-3rd-post/10230#post_12295303
 
and you should also consider if it might be an RF interference issue (see the relevant links, on that issue, below)
 

 
 

 
 
  guys sorry to ask but this volume table is for the hugo...I can't find a similar one for the mojo.
 

 
 
1/found my volume setting is blue to purple-ish (using flc 8s...have a pretty good seal, too)
based on this table i'm a bit concerned.... am i going deaf? (this is when outside, walking around, however)
 
2/also find my battery life is not 8 hrs...more like 4-5 based on my listening so far....anyfind have similar experiences?

The color setting for Hugo and Mojo is the same - the only difference is below -43 dB and above +3dB the differing light scheme kicks in. Also you can see variations in the two balls color as it gets closer to the next level.
 
As for battery life, this depends upon how well charged it is, and whether you are using USB or optical/coax, how loud you play, and the impedance of the headphones. So driving 300 ohms, using optical, green volume, you will get 8 hours. Use USB it will be 6 and a half hours. Use a low impedance IEM, green volume, and it will get worse. Use an 8 ohm loudspeaker and you will get even lower battery life.
 
Rob

 
 
Thanks to GRUMPYOLDGUY for creating the following Mojo-specific spreadsheet:
 
  The color indicator/volume problem isn't really a problem. Just start the Mojo in preset mode and count clicks to get to the right output level.
 
Preset is 3Vrms, each click is 1dB... Here's a summary of the math to get to your desired dB SPL level...
 
You need to know two parameters about your headphones:
 
Nominal sensitivity (dB SPL/mW) 
Nominal impedance (Ohms)
 
 
Step 1: Calculate sensitivity as dB SPL / 1 Vrms
Y = y0 + 10*log10(1000/Z)
y0 = headphone nominal sensitivity (dB SPL/mW), Z = headphone nominal impedance
 
Step 2: Pick a target volume (dB SPL) and figure out output level needed to achieve it
V = 10^((T-Y)/20)
V = target level (Vrms), T = target loudness (dB SPL), Y = calculated sensitivity
 
Step 3: Loss due to Mojo output impedance
Loss(dB) =  20*log10(Z/(Z+0.075))
Z = headphone nominal impedance, 0.075 = Mojo nominal output impedance
 
Step 4: Find how many dB down the target level is from the preset
L = 20*log10(V/3) - Loss(dB)
 
Simply round the number above to figure out how many clicks from the preset you need to go. 
 
I created a little spreadsheet to calculate the right levels for directly driving headphones from the Mojo... 
 

 
 

 
 
 
To set the output level to 3V ( line level ) for connection to a preamplifier press both volume buttons
together when switching on the unit. Both volume balls will illuminate light blue. This mode is not
remembered so when you switch off it will reset back to the previous volume stored for safety reasons.
 
 
 
  Has anyone experienced the Mojo keeping its 3V setting even after turning off and on again? It happened twice to me. Fist time, it almost killed my Fitear 335. Luckily I notice the huge hiss at the beginning of the song and unplugged it quickly. Second time, I saw the volume balls was illuminated in violet and had to decrease the volume (usually they are pink-ish).

 
The Mojo remembers what volume you had last and does not reset when you turn it off.


Line out mode is an exception, unless you have pressed a volume button while in it. Then the volume you set will be saved, if you did not change volume in Line out mode, it will not save to 3V for obvious reasons.

 

 
To set the output level to 1.9V RMS, first follow the above guidance, to attain 3V, and then continue further, with the following:
 
  Yes 4 clicks down will set it to 1.9v (both balls indigo). Each step is always a 1 dB change.

 
 
 
Please no worries!
However, iwas wondering if you could answer, the Mojo on line level mode - does this still run thru the Mojo's amp? from how i understand your earlier descriptions, buth the amping and DAC is done in the FPGA?
thus there is no way to truly use it as a dac without double amping?

Line level mode is just a volume preset for the volume control - nothing else changes.
 
Mojo has an FPGA (which is digital logic only) a discrete DAC (turning digital signals to analogue via flip-flops and resistors) and a single output amplifier - and that is it.
 
Conventional DAC headphone amps use differential outputs and have two I to V converters (current to voltage), a differential to single ended converter, and an output amplifier. Wrapped up with that is a analogue filter. So that's a lot of passive components and four amplifiers in the signal path. 
 
Because Mojo's FPGA has extensive digital filtering (at 2048 FS) and has a noise shaper that runs at a very high rate (104MHz) and uses a discrete DAC, I can keep the analogue section radically simpler, and this is one reason why Mojo is so transparent compared to all other DAC amps.
 
Rob

 
 
  @xtr4 i understand the FPGA designs makes the dac and amp essentally the same... what im really trying to get at is, can the FPGA's amp functions be bypassed so it is used simply as a DAC, and the two 3.5mm outs are true line outputs to prevent double amping
Paste

No, you need at least one amplifier to do the critical I to V conversion. Now it is possible to design a voltage only DAC (no amp at all), but they sound poor due to lots of problems - the largest being the huge amount of distortion you get doing it that way. Believe me, if I could make it simpler I would. The key that Mojo has is extremely low distortion and noise (0.00017% 3V 300 Ohms) but only one single amplifier in the signal path - and this amp combines headphone drive, filtering and I to V conversion in a single stage.
 
Rob

 
 
For physically-connecting Mojo to active speakers, see: www.head-fi.org/t/784602/chord-mojo-the-official-thread-please-read-the-3rd-post/13350#post_12423100
 

 
 
  I just carried a quick amp test and my results are as follows:
 
Portaphile 627x: Sadly this amp proves to be the worst offender. Mojo has out classed this amp by a large margin. 
Meir Audio quickstep: This amp did not alter the sound but to me there is no point in pairing it with quickstep as Mojo alone offers far more volume than paired with quickstep.
Wagnus Epsilon S: Expanded the soundstage which was nice but like the other amp the amp section is just nowhere as powerful as Mojo. I felt transparency also took a hit.
Analg2paper TR-07hp: This was the best pairing of the lot. Like other amps the transparency took a hit but the added bonus was the bass had a nicer reverb. To my ears the bass become extended the the decay was a lot more natural. The mid-bass to my ears was reduced and sub-bass become a little more prominent. 
 
Summary: Add an amp if you like to color the sound and play around with the tuning, I see no real value in adding amp. So far I dont have any amp that is as powerful as mojo. 

 
 
IMPORTANT: Please see the earlier section, on 'Regarding Mojos Output Stage'
 

 
This post was the stimulus for Rob's responses, quoted below:  
Quote:
 
  It is always better to give Mojo bit perfect files and let Mojo do the work, as the processing within Mojo is much more complex and sophisticated than a mobile or PC.
 
So when you have an app that has a volume control, and no bit perfect setting, then set it to full volume on the app on the assumption that this will keep the data closer to the original file.
 
The volume control function on Mojo is much more sophisticated than the PC as I employ noise shaping and I do the function at a very high internal sample rate. Hopefully using the volume set to max on the app will mean the volume coefficient is 1.0000000... so it will return the original data.
 
Rob 
  You can always do a listening test. If set to max against 50% say, and it hardens up (becomes brighter) with loud recordings then its clipping. If on the other hand the perception of sound-stage depth is reduced, then the volume control is degrading the sound.
 
If you do that test and can hear no difference then don't worry, its a good app volume control.
 
Rob
 
PS for fun I just did a very quick test using Dave. I listen to radio 3 using the BBC iPlayer. I normally have it set to max. I reduced the iPlayer volume control to half, boosted Dave volume control by +6dB - and yes I felt sound-stage depth was worse with lower iPlayer volume.

 
 
IMPORTANT!: Android automatically upsamples ALL music files to 24/192, which is not a good thing for Mojo - here's what to do about it
IMPORTANT!: please also see discussion in the 'Connecting Mojo to OTG (microUSB) devices' section, above)
 
 

 
Please note: Instructions advise charging a brand-new Mojo for 10 hours before using, but actually, it is only necessary to charge until the tiny white charging LED goes out. With most brand-new Mojos, this will be around half that. Just trust what the charging LED tells you.
 
 
For a Mojo that has already been charged previously:
 
Quote:
Originally Posted by Rob Watts 
 
Charging times - its 4 to 5 hours if the unit is off from flashing red to full charge. But if you are using it at the same time, it will take much longer (maybe 12 hours), as current is being drawn to feed Mojo and less to charge the battery. Check that the charging LED is not flashing, as this indicates a fault such as insufficient current from the PSU.
 
Rob

 
 
If your Mojo LED is flashing whilst connected to a charger, please check that the charger is rated at least 1amp current-delivery
 
 
Quote:
 
@Rob Watts

Just a quick question. Ive been using my Mojo just below the double red volume. Got 4hrs out of it and the battery indicator went yellow. Does this seem about right? Feels like i may only get 8hrs from a full charge at well below my normal listening level. Im assuming harder to drive headphones may get 5+hrs out of a charge. This seems lower then i expected.

I have dug out my original design notes and measurements from the battery ADC built into the Xilinx FPGA from one of the prototypes.
 
The intended colours for battery life are:
 
Blue              100% to 80%
Green             79% to 50%
Yellow            49% to 10%
Red                  9% to 2%
flashing red     less than 2% or 10 minutes left.
 
Use this as a rough guide only, as the battery voltage and life left was not exact!
 
Rob

 
Quote:
Originally Posted by Rob Watts /img/forum/go_quote.gif
 
.... when battery is fully charged even after 10 hours or 5 led indicator is automatically turn off, when I put it to charge after charging the led indicator turn on again ( white color ) and looks like battery never been charged .


Don't worry that's fine, the charger has been reset and its in trickle charge mode.
The colour without the charger connected is the one to watch (fully charged is blue then green, yellow down to red, flashing red means 10 minutes left).
 
Rob
 
 
First, check that your charger is rated for at least 1amp charging current (higher is fine; lower is not). If the charger is not rated high-enough, then Mojos white charging LED will flash, to warn that Mojo will not charge successfully.
 
However, if your charger is fine, then it may be that Mojos battery has been discharged more-deeply than usual:
 
Quote:
 
  .... I am thinking the battery is not holding any charge. 

Try charging overnight with the unit off. The charging circuit looks at the state of the battery before charging. If the battery has a very low voltage, it will trickle charge the battery until it gets to a safe voltage, and then full charge will commence. This trickle charge mode can take several hours, and it is done for safety reasons, and it will appear that the battery is not working as the trickle charge mode takes some time. When in this mode Mojo must be off.
 
When charging make sure the battery light is white and not flashing - if it flashes, pull out the charging cable, count to ten, re-attach the charging cable. If it continues to flash, it is most likely the charger is not giving 1A at 5V, so use a better charger.
 
Rob 

 
Originally Posted by Rob Watts
 
 
Firstly the 4/5 hours is the charge time whilst it is in constant current or full charge mode - so that will get you to blue. But after that it goes into trickle charge mode, and the white light will still be on. I can't remember how long the trickle charge mode is, but I guess 9 hours would be right for full and trickle charge.
 
Of course, if you are charging whilst on it will take very much longer to charge, and the charger timer might get triggered then you get the flashing white battery LED. The charger timer circuit is only on during full charge mode. So if it's fully charged, and then you plug in the charger and turn it on, then the white charger light will stay on permanently as the trickle charge is being balanced by the current Mojo is drawing (no net current into or out of the battery).
 
Hope that explains!
 
Rob

 
 
 
It can push 5.3V when fully charged.


Charging state of the battery makes little difference to the output level and mojo actually has no problem in driving HD 800's by our precise and accurate measurements. There are no significant or measurable changes to the output with these headphones so possibly it's something else happening here. Interesting though!

 
 
Quote:
 
Originally Posted by Mojo ideas /img/forum/go_quote.gif
Quote:
I am thinking of buying a Mojo (already have a Hugo) as it is more portable than the Hugo and easier to charge (no need for the specific charger)

One question I have for Chord is how many charge/discharge cycles is the Mojo battery going to last ? I would expect 1000 cycles at least before 80 % capacity is used.

Anyway the Mojo is not a mobile phone that needs to be on 24 hours a day anyway.

And is this battery use replaceable in 5 years time?

Thanks

Well in excess of 2000 however that is quoted as full deep discharge so any lesser level of discharge counts towards a full one so you'd need two 50 percent cycles to count as one full one this means that as our battery is unlikely to undergo repetive deep cycles it's life is calculated to be out beyond 12 years . The battery is a brand new advanced design. It can be replaced easily in any-case as it has a plug in connector,
 
Quote:
Originally Posted by Mojo ideas /img/forum/go_quote.gif
.... the batteries are expected to last far longer than three years. The batteries in our designs are not subject to damaging deep discharge cycles or anything more than very light current demand .... Batteries used for power tools are quite a different matter, but in our units expect a life of greater than ten years. Mojo has a plug on it so it's just an easy replacement for a shop technician Hugo batteries will need soldering in place though but this is also a low skilled job which would be require rarely if ever.
John Franks.

Quote:
Originally Posted by Rob Watts /img/forum/go_quote.gif
The batteries are plug in and held into place via thermal sheet. Very easy to change by your dealer.
 
You should see more than 10,000 hours of use before the battery will need changing.
 
Rob
 
We had the battery developed for only our mojo application. Done for us especially, It took Chord 3 years and many attempts to get the sheer ear thumping power density we have achieved in mojo. So I'd rather people didnt underestimate our design skills and I'd ask please don't think you can better it with a quick battery substitution as this can be risky or even dangerous.

 
Quote:
Guys we at Chord really did extensive studies into batteries before we chose the optimised solution we have in Mojo. We even looked at up and coming battery technologies like lithium sulphur which potentially could extend Mojo's playing time by a factor of four but unfortunately the newer chemistry's are just not there yet. Be aware the battery technology we've chosen is good and above all its safe being a higher spec than most. Remember that mojo is designed for a pocket or a hand to carry and a poorly deigned lithium ion battery could be chemically very volatile. Look up lithium batteries catching fire I don't want one of those in my pocket!

 
 
 
There is a minute amount of battery usage when the unit is turned off as it has to monitor the button states. So it can be expected that it will need a small charge after not being used and off, this shouldn't be a problem as it will be a few months before mojo loses all charge when switched off.
 
With regards to the 3 buttons flashing, this is not a problem when the unit is charging. 

Mojo will discharge the battery - it will take about 6 months to do. But after finishing a charge, if you reconnect a bit later it will re-charge with the white light on. But it is only supplying a few milli-amps of current, as the last part of the charge is a trickle charge.
 
So the white light on is nothing to worry about.

Originally Posted by Rob Watts

 
the battery is always connected - but - when the charger is on, and the battery is fully charged, then the trickle charge is balanced by the current that the amp needs, so no nett charge going into the battery - its just going from the charger to the amp... The battery is still providing a low impedance, and dynamic surge currents though, but the average DC current is just matched by the charger.
 
Rob 

 
 

 
Is there any harm in most of the time leaving the Mojo hooked up to computer and plugged in to the wall to keep a full charge?

No it's fine to leave it plugged in all the time but if your charging from an unplugged lap top you may drain the lap tops batteries. But if your using just the data USB connection ithe mojo takes no power from the connected device. John F.

 
Quote:
 
Mojo, like Hugo, has been designed so that you can have the charger plugged in constantly. So on a desktop charge and run it at the same time. Once its fully charged, the charger will just supply enough current to balance Mojo's current draw, so no net current from the battery.
 
Rob

 
Quote:
 
Just to clarify. Charging is automatic. If you are playing and charging at the same time, with a fully charged battery, the charger will supply enough current to balance the consumption used by Mojo, so no net current into the battery. If its fully charged and the unit is off, the charger will go off. The charger will re charge automatically when the battery voltage falls to 8.2v (off at 8.4v) so keeping the charger connected will ensure a full charge.
 
Rob 

 
Quote:
 
  I am also considering replacing my desktop DAC (which, by the way, cost far more than than the Mojo) with a second Mojo, so I might have something to sell too! One question that I think remains unanswered is, Is it alright to leave the Mojo on 24/7 plugged in with a 2A wall wart? Will that adversely affect the battery or anything else in the Mojo? Would keeping the Mojo on all the time avoid the issues with the battery charging circuitry?

It was designed to run this way. If you want to maximize battery life, then turn Mojo off when not using it, with the charger connected permanently and it will be fine as the charger will disconnect automatically, and re-charge automatically when the battery voltage drops.
 
Rob

 
 
 
Quote:
 
Hi Rob, and John,

1. If I leave the Mojo plugged in to the wall wart all the time, that should be fine, correct?
2. I presume that any time I want to resume listening in its present setup (meaning listening via the computer and with the charging cable still connected to the wall wart), the Mojo should be "ready to go" and that there is no need for me to ascertain the battery level? 
3. If I then use the Mojo as a portable, and later on connect it back to the computer for listening (with the charging cable then connected to the wall wart again), what would be the minimum color of the battery light? Green?

 
1) yes it should be fine
2) overall there is a slight net drain on the battery so starting at blue or green on the battery indicator is a good idea but not mandatory
3) difficult to answer as it depends how long you've been listening whilst mobile but see answer above. Happy listening George!

 

John answered 1 and 2 fine, but I thought I would clarify exactly what happens when you charge and listen at the same time.
 
I use a dedicated charging chip for the Li battery, and this has a number of safety features, and works with a number of settings to ensure safety.
 
Now one of the safety circuits is a safety timer, and this is when the charger is in full charge mode. This timer is set to about 8 hours, and normally full charge mode takes 4 hours, when the unit is off. But when the unit is on and playing, there is a risk that the safety timer will be set, as it can take 12 hours to fully charge (from flashing red) and when playing music (for those 12 hours) at the same time. If the safety timer is set, then the battery LED will slowly flash white, and no further charging will take place. To reset the timer, just disconnect the charge USB, wait 10 seconds, and reconnect, and it will recommence charging. So if you are charging and playing, then when you have finished listening, turn Mojo off, and it will be OK. When Mojo is blue, and you connect the charger, then it is trickle charge mode, and the safety charger is not operating. So if Mojo is green, the safety timer won't trip out, as it will play and leave full charge mode within 8 hours, so you will be OK. It should be OK at yellow too. I guess the easiest way of dealing with it is to turn Mojo off after listening, then you will be fine, unless you listen for longer than 8 hours starting from fully flat.
 
Note that you can get the flashing battery LED if the USB charger voltage is low, from a charger that can't supply the current, or a USB cable that has high resistance. But you will see this pretty early on.
 
I hope this clarifies.
 
Rob

 
Quote:
Originally Posted by Rob Watts

 
the battery is always connected - but - when the charger is on, and the battery is fully charged, then the trickle charge is balanced by the current that the amp needs, so no nett charge going into the battery - its just going from the charger to the amp... The battery is still providing a low impedance, and dynamic surge currents though, but the average DC current is just matched by the charger.
 
Rob 

 
Quote:
  If you fully charge Mojo then use it in a desktop it will not switch off; the power dissipation that the charger uses in matching the current drawn by Mojo is negligible. You are only at risk when charging and using at red ....
 
Just to give you some numbers - fully charged and matching Mojo's current draw the power dissipation is 107 mW for the charger circuit. That will increase running temperature by less than 1 deg C. But at flashing red it is 910 mW for the power dissipation in the charger.
 
Now I could fix this by using a switcher based charger rather than a linear one - but these inject too much RF noise onto the battery. This would impair sound quality, and Mojo's design goals was that plugging in the charger would have no significant change in SQ - which would not happen if I used a switcher based charger. I am not prepared to damage SQ as to me this is the most important aspect just for a tiny improvement in usability.
 
Rob

  To understand it better, let's assume Mojo is off and charging.
 
Now the charger has two modes of normal operation - constant current, which is set to 330 mA, and constant voltage which is set to 8.200 V. Now when the non charging battery battery voltage is less than 8.200 V, then the charger supplies a constant current. But when the non charging battery voltage gets close to 8.200 V, then the charger switches mode to constant voltage at 8.200 V. The current that is charging the battery then slowly falls from the initial 330mA, to zero - its in the trickle charge mode now. Eventually, the non charging battery voltage hits exactly 8.200 V, the charger is in constant voltage mode of 8.200 V, no current now flows into the battery, and the charger switches off automatically. When the battery voltage falls to 8.0 volts, then the charger will return to charging. Tip - if you want to force the charger to top up Mojo's battery to 8.200 V then removing the charge USB, wait 5 seconds, reattach, and the charger will top it up to 8.200 V.
 
Now imagine that Mojo is on at the same time as it is charging. In this case, the battery will continue to charge until it gets to 8.200 V, and the charger is set to voltage mode and gives 8.200 V too; so no current flows into or out of the battery; but Mojo itself is drawing 180 mA of DC current, and this will simply come from the charger - so the charger will supply the needed 180 mA for Mojo. It will do this for ever, and it won't switch off. This is intended, as it means that the battery is effectively not being used to supply the bulk of the current, won't charge or discharge, is held at a safe level, and will operate like this for a very long time.
 
Now we have been talking about DC currents, and this is indeed the vast bulk of the current. But what about dynamic currents and noise? Because the output impedance of the battery is much lower than the charger, then the noise of the charger is reduced; also dynamic currents still comes from the battery. So running in this mode ensures the best of both worlds - low RF and audio band noise from the battery, large dynamic currents available, and low PSU impedance too - but without the worry of the battery wearing out from charge and discharge cycles.
 
I hope this clarifies.
 
Rob 

 
Quote:
when the Mojo is being used and being charged especially when driving lower impedances there is a net drain on the battery. This means that the charging circuit does not quite provide enough power to power the Dac and amp circuitry and keep charge the battery at the same level this is because we had to limit the amount of charge over a given time due to thermal constraints. Our charging time with the Mojo switched off is usually four or max five hours this is a little inconvenient but when we compare this to other Dac amps that need up to a full twenty four hours to charge we feel that we didn't do such a bad job.

 
 
 
 
Broadly-speaking, most people get around 7-8 hrs from a fully-charged Mojo, but it can vary depending on, for example, what load your IEMs/CIEMs/Headphones present to Mojos output stage, how loudly you play your music, and also (to a small degree) what digital protocol you are using:
 
low load, opt/coax may yield closer to 8 hours
low load, USB may yield closer to 7 hours
 
heavily loaded then you could lose another hour.
 
Low load would be -20dB FS into 300 ohms, 3v preset volume.
 
 
 
Quote:
  As for battery life, this depends upon how well charged it is, and whether you are using USB or optical/coax, how loud you play, and the impedance of the headphones. So driving 300 ohms, using optical, green volume, you will get 8 hours. Use USB it will be 6 and a half hours. Use a low impedance IEM, green volume, and it will get worse. Use an 8 ohm loudspeaker and you will get even lower battery life.
 
Rob

 
Quote:
   
 optical is the lowest power - the USB decoder chip is about 1/3 W and is turned off when VBUS is low.
 
Rob
 
Quote:
  .... the impedance of headphones can be reactive, so there could be a phase shift between current and voltage, thus slightly increasing power drain. But don't worry about that. The max power being drawn within Mojo is with 8 ohm IEM's - as although the power in the load is low, the current is higher and that will increase power dissipation on Mojo's discrete OP stage, even when you are running say at red on the volume.
 
Oh another point - power loss within the battery is 0.2% of Mojo's total, so it is insignificant.
 
And another one - power loss whilst charging is because of a use of a linear charger - so when the battery is fully depleted, we get max power loss in the charger, and very little power loss whilst at the end of the charge cycle. Why do I use a linear charger and not a switcher based charger? I have yet to find a switcher charger that allows full RF filtering, so it will upset the sound quality whilst charging. My design goals were to have no loss in sound quality whilst listening and charging and going to the current range of switcher chargers won't do that.
 
Should you find the thermal trip operating whilst charging and listening, then as a poster recommended, putting Mojo on its side fixes that possibility.
 
Rob

 
 
 
Originally Posted by Rob Watts

Does it do any damage if one inadvertently puts power into the micro USB data port?
 
Not if it is a legal VBUS +5v. Now Mojo is protected for over-voltage on VBUS, to prevent possible failure of the battery charger; this protection will fail if too much voltage and current were on VBUS; if the protection diode fails it will short, thus ensuring that dangerous over-voltage will never damage the battery charger; a failed protection diode would then need to be replaced. That said, I am not aware of any failure of protection diodes at all.
 
Rob 

 
 
 
Quote:
 
 
  I found the reverse.  I'm using Sennheiser HD-25 1 II: directly out of the Mojo the sound seems present and correct, but when used with a Ray Samuels SR-71a, the sound goes to a whole new level.  The sound becomes rock solid and more like listening to musicians playing instruments; without the Ray Samuels the sound seem to collapse in on itself and become more hi-fi (ie impressive noises but less music).  To my ears, the extra amplification is not adding tonal euphony but is instead making the most of the DAC.
 
I have a theory that it's to do with the power supply: when using headphones more current is drawn and in a varying manner, ie it varies with the music.  This varying of current affects (I think modulates) the power supply voltage which affects the DAC, amplification and ultimately the sound. By connecting directly to an amp, there is less current drawn and no variation.  This might also explain why companies such as Naim claim improvements to their amps' sound quality when external power supplies are added.  Just my 2CW.


I do not buy this all. You need to bear in mind several facts:
 
1. The battery is capable of delivering 3A of current, and has very low impedance.
2. Mojo amplifier has a very high power supply rejection ratio.
3. The output is pure class A at 5v RMS into 300 ohm.
4. Reducing the output load only starts to increase distortion with 33 ohms - at this level it is very much lower than other headphone amps. The HD25 is a very easy 70 ohms.
5. Mojo is designed to drive loudspeakers. You will be amazed hearing it fill the room with beautiful sound using efficient 8 ohm horn loudspeakers.
 
Some people like the sound of more distortion - 2nd harmonic fattens the sound making everything sound phat, soft and rounded. But its not natural, nor do I find it musical, as everything sounds phat. I want soft sounds to sound soft, and sharp sounds to sound sharp - not everything to have a soft sheen on things all the time.
 
I can give you another example. I just had an email today from a very experienced dealer that asked me this question:
 
"Chord Mojo should have single amplification which drive to both headphone output, but if I'm using any headphones (HD800 for example, very heavy to drive) to put on headphone output 1 and I connect another headphone to headphone output 2 (Beyer T1 for example), I hear no differences on sound quality. Normally if one amplification section used to drive 2 headphones output, then once we connect the second headphone will make overall sound quality degradation (as the case of Beyerdynamic A20 amp, Grace Design M903, etc). What is the logical explanation of this? As it seems Mojo has 2 separate amplification sections which drive independently for each headphone output."
 
Of course Mojo does not have two amplification stages. It can drive two headphones with ease because it has exceptional low output impedance, and it has exceptional current linearity. So loading it with more headphones has no effect, unlike other headphone amplifiers.
 
Indeed, when I initially started designing Hugo I was shocked how poor from a measurement point of view headphone amps were. Poor output impedance, huge levels of distortion, and poor current linearity seem to be typical. And these things matter, if your goal is transparency and musicality. 
 
Rob
 
Quote:
  The earlier prototypes had less current than Hugo - 0.2A RMS. But last Feb we were with Nelson (Malaysian distributor) and I showed him the prototype. He brought some headphones, and we compared it to Hugo - and it was bad, a lot of clipping from Mojo, none from Hugo. It so happened (luckily) Nelson gave us possibly the only 8 ohm headphones on the market Final Audio Pandora.
 
So I upgraded the current to match Hugo, its the same 0.5A RMS. To do this I used 6 small OP transistors in parallel as size would not permit use of the large devices I use in Hugo.
 
Mojo development had some weird fortuitous events - the first time we showed the prototype happened to use the only 8 ohm headphones that would show problems - and the day we were deciding on the design Xilinx emailed me with a new FPGA that would enable Hugo performance but with the needed power, and happened to be in production just in time for Mojo. 
 
 
  Does anybody know the battery capacity on the Mojo? sometimes i use a 5000mah external battery, which the Mojo completely drains, and it barely goes to green level(i get about 4--5 hours of use after this charge). Also, i read that Chord recommends charging from a 1A/5V source. Is there any problem if i charge with a 2A/5V iPad charger? Honestly i didn't check to see if it shortens the charging time. 
The thing is i travel a lot, and i prefer to have with me a single 2A charger, which i can use for my iPhone, iPad, and hopefully for the Mojo.

Mojo's battery is actually about 14Wh - Watt hours - is a better measure of battery capacity, as Mojo has two cells with a max total voltage of 8.4v. Your 5000mAh external battery is maybe only 18.5 Wh (assuming it is a single cell), and when you figure in inefficiencies in power delivery, you will need more than 18.5 Wh from a portable battery to fully recharge Mojo. 10000mAh single cell should be fine.
 
As other posters have said, 1A is min, 2A is fine but Mojo will not charge any faster. I have controlled the charge time for thermal reasons.
 
Rob

 
 
Quote:
JF here - Mojos multiple DSP cores and all other circuitry develop 1.7 Watts of heat when running this heat it dissipated from Mojos case through convection and heat radiating away. This can only happen when the Mojo cases temperature is a few degrees above the ambient temperature so it will feel warm in a high ambient environment. This is normal and totally safe as there are three separate and independent thermal sensing and protection circuits to look after Mojo and Mojos special battery.

 
JF here - ....  the battery is perfectly safe right up to 150 degrees Centigrade we had it made that way and its costs more than other batteries. The case of the mojo when it's charging has to shed about 1. 7 watts of heat it can only do that by radiating it away or convecting it away. If it's in a warm environment or it can't covect its heat away it's temperature will rise until there is a sufficient temperature differential to shed its heat. It's perfectly safe we have three indepentdant thermal temperature safety shut down method in mojo so please don't worry we know what we are doing. Remember a really hot cup of tea is usually only about 60 degrees C.

 
Mojo actually has three independant thermal cut outs a special high temperature battery and very sophisticated charging circuitry . Picking up on an earlier post Mojo actually does not dissipate a lot of heat when it's working. It's only about 1.7 watts and when it's charging it adds about another watt so its not much really. However the electronics and battery are thermally bonded to the aluminium case. The Mojo's case can only shed its heat through convection or by radiating it away. This can only work if there is a temperature differential between itself and its ambient surroundings if there is an insufficient gradient between them, the Mojos temperature will rise until there is a large enough difference to pass its heat to the air surrounding it.
If it is prevented from doing this perhaps by being insulated I some way it's temperature will rise until one of the three shut down trips operate. note the battery is safe to 150 degrees and the trips all operate up to a hundred degrees lower. Therefore it's perfectly safe. In fact if it's feeling mildly hot at first to your hand. Your hand alone will easily soon bring the unit down to a reasonable temperature.

 
 
   
.... The unit turns off after a while. Probably when it gets too warm. That was kind of a disappointment as I was going to use it as a DAC for my PC as well.

You get the most power loss when it is charged from red; and when its being charged at full blue then the power from charging is very small.
 
If you do need to charge & listen at the same time from red, and its in a hot room, then if you charge it with Mojo on its side so the top & bottom is in free air, it will not turn off. A head-fi poster mentioned this & it works well, as Mojo's power dissipation is almost doubled by doing it this way.
 
Rob

 
Quote:
  If you fully charge Mojo then use it in a desktop it will not switch off; the power dissipation that the charger uses in matching the current drawn by Mojo is negligible. You are only at risk when charging & using at red - & indeed as Mython says putting Mojo on its side will solve that issue too.
 
Just to give you some numbers - fully charged and matching Mojo's current draw the power dissipation is 107 mW for the charger circuit. That will increase running temperature by less than 1 deg C. But at flashing red it is 910 mW for the power dissipation in the charger.
 
Now I could fix this by using a switcher based charger rather than a linear one - but these inject too much RF noise onto the battery. This would impair sound quality, & Mojo's design goals was that plugging in the charger would have no significant change in SQ - which would not happen if I used a switcher based charger. I am not prepared to damage SQ as to me this is the most important aspect just for a tiny improvement in usability.
 
Rob
 

 
Quote:
Originally Posted by Rob Watts /img/forum/go_quote.gif
 
The noise is due to ripple voltage on the charger upsetting the inductors/capacitors within Mojo. If you use a clean quality PSU & a low resistance USB cable to the charger PSU the mechanical noise should be silent or insignificant. 
 
Rob

 
Quote:
Originally Posted by Rob Watts /img/forum/go_quote.gif
 
The problem is a noisy USB VBUS power line, & this makes the inductors in Mojo vibrate. Also, some USB cables have high resistance, & this makes the problem worse - so using a different cable can make the mechanical noise go away. The PSU itself can make it better or worse. Don't worry about it if you hear the noise, Mojo is not faulty & will continue to be reliable.
 
Rob

 
 
  is it normal to have the mojo make a hissing sound (ie the unit itself, audio output is fine) when plugged in to power? brand new unit but it makes this weird noise when its plugged in (both when off and on). the hissing dies down when i plug in the signal cable (unit still off), not sure if i should send my unit back.

It's normal - it is the charging regulator going into low power mode. Don't worry, there is nothing wrong.
 
Rob

 
Further testing on the charger and cable compatibility. Chosen Anker PowerPort 5 for good measured performance and multi-port charging with Anker PowerLine due to good construction with braid + foil shielding. All are available at a reasonable price.


Case 1: Virtually silent, only heard very minor hiss when ear is pretty much on the unit
Case 2: Only noticeable hiss when you put your ears near the unit
Case 3: Loud whine lasting the only the first few seconds, faint charging noise after that.
Case 4: Loud whine, and it goes on for a few seconds and off for a few (voltage drop causing charging circuit to shut down)




Combo 1: Sony or Samsung charger + Sony or Samsung USB cable + extension cord = Case 4

Combo 2: Sony or Samsung charger + Any cable = Case 3

Combo 3: Apple 1A charger + Sony or Samsung USB cable = Case 2

Combo 4: Anker charger + Sony or Samsung USB cable = Case 2, slightly quieter than combo 3

Combo 5: Sony or Samsung charger + long 6ft Anker cable (same cable length as Combo 1) = Case 3

Combo 6: Apple 1A charger + Anker cable/Chord Mojo's bundled cable = Case 1

Combo 7: Anker charger + Anker cable/Chord Mojo's bundled cable = Case 1, slightly quieter than combo 6 directly compared, with no pattern to the noise (always the same loudness)



Anker PowerPort is tested at it's worse case scenario(unloaded), the ripple and spike measurements are better when the charger is fully loaded with devices. The Anker charger have noticeable more steady noise pattern than the Apple charger even with the best cable connected, the Apple charger's noise ripples in loudness and the Anker one is very steady at the same amplitude.



In short:

If you already have a Apple charger handy, just getting a quality USB cable like the one I've tested will yield noticeable gain, especially if you are using longer cables.

If you don't have an Apple charger, getting an Anker charger with their PowerLine cables will yield the best possible result without going to go with a lab bench linear power supply. Having a multiport desktop charager will also allow you to run shorter cables.

Link where I bought them:
Anker PowerPort 5: http://www.amazon.co.uk/dp/B00VTI8K9K
Anker PowerLine: http://www.amazon.co.uk/dp/B014H3GKZ4


 
 
 

 
 
Interference is only noticeable when your cell is on 2G, & only if your IEM cable is within 3-6 inches of the phone (that seems to be where the interference comes from, the headphone jack)...

3G/4G/LTE etc do not have any effect that I can tell...

 
Quote:
Waaaaay back, early in the thread it was determined that 3G and LTE showed little to no EMI noise, but 2G/Edge cellular reception was very noisy. I heard no noise until I switched to 2G/Edge & it was brutally obvious. That's another factor besides cables acting as antennae.

 
  ok seeing everyone has earned a day off at The Pump it's down to me to bring you a 'Official' Chord-Electronics announcement dealing with RF interference with phones...
 
We at Chord Electronics suggest that people switch to flight mode when using Mojo Especially when it's strapped to a phone.
 
When using a phone as the source the RF noise level is dependant which frequency its on & how far the base station is away from the phone.
 
This is because the phone will ramp up its transmit power to make the connection.  This is almost like an EMP bomb going off right next to the Mojo and its two dangling RF receiving cables. We have taken all precautions, but there is little that can be done to overcome the massive amount of RF that is generated within millimetres of the Mojo, hope this helps in some way. 

 
  The problem lies in the cables that we attach both input & output & the variations thereof. These act as aerials feeding directly into Mojo. A phones level RFI in close proximity to mojo is very severe & therefore this issue is not easily solved without compromising Mojos performance this is because when a phone loses signal it ramps up the transmitt levels dramatically & these can be on any number of frequencies. Some cables are adequately screened & with those there is unlikely to be a problem, but with unscreened types there may be. That is why we recomend that for critical listening & in environments where a signal is likely to be lost that you switch to airplane mode.

 
  You can get various sizes of snap-to-fit ferrites at most electronics shops.
https://www.radioshack.com/products/radioshack-snap-choke-core?variant=5717355973
 
I'm not saying it will make a difference to the sound quality etc.

 
   
Basically, if you experience RF issues when Mojo is connected to a smartphone, then:
 
  1. ensure you are using a proper coaxially-shielded connector cable
  2. if available to you, try to operate your smartphone on 3G or LTE, & not on 2G/Edge
  3. try using the smartphone in 'Airplane' mode whenever possible
  4. Co-ax cables, & USB connector cables, can often be purchased with a ferrite choke manufactured integral to the lead, but if your cable does not include one, you may find it worthwhile obtaining a small ferrite choke to clip around the cable, locating it as close to the DAC end as possible
 
  1. Quote:
  We are hearing good things about Audio Quest jitter bug - I have heard it stops the mobile phone EMC problems that can happen with Mojo & certain headphones. I will be checking it out soon.
 
Rob
 

 
Where can I BUY Mojo in my Country?
 
 
Where can I BUY Mojo in the UK?
 
 
Where can I REGISTER my Mojo product?
 

 
Oct 14, 2015 at 8:56 AM Post #4 of 42,758












  1. Animated rotating view of Mojo (Click to show)







Peter Hyatt's - 'Mojo's Greatest Hits' (music recommendation page for tracks that sound excellent through Mojo)






Since I can no longer edit the first 3 posts on this page (due to restrictions that were imposed by the changed forum software-platform), here are some assorted recent posts, of potential interest to Mojo customers, but in no particular order:


Hugo 1 and Mojo both have similar depth performance - not surprising, as depth is down to small signal amplitude linearity, and Hugo 1 and Mojo have exactly the same 4e pulse array, noise shapers and internal truncators - and it is these items that determines small signal linearity and hence depth perception.


A DAC has two parts - the digital, and the analogue. Conventional DAC chips use a mixed signal process, so it combines digital and analogue, but typically would be in a 180 nm process. So the DAC designer has to keep the digital parts very simple, as there is not much area and so few gates to do anything complex. Moreover, the digital part is noisy, and upsets the analogue part through the substrate. Also, analogue on silicon is a big problem - resistors and capacitors are non-linear, so one has to go to very complex lengths to reduce this problem.

But an FPGA based DAC is actually not a DAC; what happens is the digital part is on an FPGA (which is a field programmable gate array) and the analogue part is via discrete analogue components. The beauty of this approach is that you can have an extremely complex digital part, as the FPGA is made with 28 nm silicon, so you can pack many more gates in an economic device; Mojo has 500 times more processing power than usual chips because of using an FPGA. Also, because the analogue part is discrete, there are no issues with noise coupling, and resistors and caps are all linear. When I have desiged the digital parts for a chip, it is fundamentally the same process as designing for a FPGA; indeed, I always prototyped my silicon chip designs with FPGA's.

But there are downsides to an FPGA DAC; the designer must know what he is doing; and unit costs are very much higher - but that doesn't matter too much for high end audio, where performance is the most important factor.


No that is not the situation, we did not balls up RFI.

Chord and I have been working very hard on this issue, and spent a lot of time and effort trying to improve the RFI. So the first mod improved Wi-Fi, and this went into the production model. But people kept complaining, so I tried iss3, with RF improvements.

Iss 3 had no benefit.

So I tried iss 4.
This had no benefit.

Iss 5 - this too had no benefit.

At this stage I had tried everything conceivable to improve RFI, so I handed the design to a RF engineer, who builds RF OP stages for mobiles.
He suggested some changes.
I took his ideas on board, and actually improved upon them, with filters that knocked out every single mobile phone freq across the World. And then I beefed it up with three of these filters.

So I was now convinced it would kill the problem.

But - iss 6 had NO BENEFIT again - what was alarming was it made no difference whatsoever.

But then realisation occurred: the RFI problem IS NOT RFI.

Why did I come to this conclusion? Because RF treatment has no effect at all, but people were claiming that mild steel helped. Mild steel has no screening effect on RF, but screens for low frequency magnetic problems. The RFI issue is in fact the phone having poor low frequency magnetic fields as the RF is modulated. This explains why most phones have no problems whatsoever, but other phones have big issues, when the actual RF transmission power is identical.

So to help with the issue you can:

1. Get a better phone with low external magnetic fields.
2. Use a case that has mild steel plates on the surface between Mojo and the phone.
3. Keep the mobile some distance away from Mojo.

Note that this issue is not just a Chord issue, as this noise affects other DAC amps too. Note too that it's not an issue with all phones - I have personally never ever heard RFI on Mojo or Hugo 2.


Windows 10 drivers. The full update process is Win 10 driver, then install the Win 10 Creators Edition on top (both available from the Chord website). There were a few issues, but I think those were due to either owners not installing the Chord driver, or installing the Win CE driver without installing the Win 10 driver first. I don't remember any posts about Win 10 drivers for quite a while, so I assume that the drivers are now stable and mature, and no longer causing issues.

So some guys from JRiver helped a bit. It is not possible to play DSD files or have the Mojo convert PCM to DSD and have windows sound at the same time.
Here's the thread and explanation https://yabb.jriver.com/interact/index.php/topic,114282.0.html

Chord also says it is not possible.



A Hugo 2 post, possibly of interest to Mojo owners,, but unrelated to the above:

Just a note.

Tonight I was trying to launch a PC game and it would not launch. Neither would two other games. Then an app called Desktop OK would not launch.

I found the problem. It was because I was listening to the Hugo 2 on optical input from another source, rather than my computer. However I was idly connected via USB between my Hugo 2 and computer. This was somehow causing programs to fail to launch.
The moment I turned my Hugo 2 off optical, and put it onto USB, all programs launched as normal.



With portable devices it is all about power, from two POV - battery life, and how warm the unit gets. So a bigger case = bigger more capable batteries, and larger surface area to dissipate the heat. So Mojo's FPGA has half the power than Hugo 2's - even though it's the same FPGA - as pretty much with Hugo 2 I am using max capacity and max internal clock speeds. And I can increase the element count on Hugo 2, as more power is available. So when using it as a DAC, the improvements are mostly down to the FPGA and the 10e pulse array.

Now the OP stage design is quite different in that Hugo 2 uses the method I created for Dave - the 2nd order analogue noise shaper - and this has big benefits particularly when using low impedance loads, such as anything below say 100 ohms. Distortion with low loads does not significantly increase with Hugo 2, and this is down to the analogue noise shaper topology. So the benefits of this will depend upon the headphone used.




Another CCK-circumvention cable option (not actually confirmed in this thread, but looks worth a try, considering the reasonable price)

and Another CCK-circumvention cable option (note: this suggestion is in regard to the updated version of the Meenova cable; not the old version):

Just an update for the newer Meenova cable - has been working without issue over the past 24 hours. I've spent at least a few hours listening, plugged/unplugged a few times. It's so nice to be able to use my iPhone w/ Mojo again.

www.meenova.com/st/p/lgtnmuc.html

(IMPORTANT: Meenova state that this is only functions with iOS 11.2.6 or less)
 
Last edited:
Oct 14, 2015 at 9:02 AM Post #5 of 42,758
This is very exciting! I'll be heading up the Canadian tour and will have the thread up later today as soon as I can. Filled.

Expected price in Canada is $799 CAD.
 
Oct 14, 2015 at 9:02 AM Post #6 of 42,758
Great stuff! Will be posting here. :^)
 
Oct 14, 2015 at 9:11 AM Post #8 of 42,758
http://www.digitalaudioreview.net/2015/10/chord-electronics-mojo-portable-audios-new-talisman/
 
Oct 14, 2015 at 9:14 AM Post #9 of 42,758
Okay, I'll sacrifice myself and just quickly post this here this one time, so we can be done with it once and for all.
 
https://www.youtube.com/watch?v=a6Acigj8isc
 
May this stand in lieu of all the future jokes that will hopefully never be.
 
Oct 14, 2015 at 9:23 AM Post #11 of 42,758
As a perfectly happy 901 & minibox owner, I am going to sit back and watch this one unravel.

Only interested in one box portability, and my Mdac at home isn't worried at all.! I hope :blush:
 
Oct 14, 2015 at 9:26 AM Post #13 of 42,758
Just picked mine up from Custom-cables. Guys this thing is tiny! The picture make it look a lot bigger than it is. More pictures and details later. I also got the Hugo plus the Oppo HA2 so would compare later.
 
Oct 14, 2015 at 9:26 AM Post #15 of 42,758
As a perfectly happy 901 & minibox owner, I am going to sit back and watch this one unravel.

Only interested in one box portability, and my Mdac at home isn't worried at all.! I hope :blush:


Hugo like sound at a much more accessible price..........
 

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