[Arpiben]

   
I don't mind higher output power and different electronics components at all as long as the signal path remains as direct as it is in the original Hugo.

Same here

.In theory,the most direct path will be preferable with one or two OPs ' (as Hugo/Mojo/Dave if I am not wrong) instead of 3 or 4 Ops in a row.
Rgds. 

 

[JaZZ]

 

  I don't mind higher output power and different electronics components at all as long as the signal path remains as direct as it is in the original Hugo.

Same here

.In theory,the most direct path will be preferable with one or two OPs ' (as Hugo/Mojo/Dave if I am not wrong) instead of 3 or 4 Ops in a row.
Rgds. 

 
What do you mean by «OP»? This acronym was used by Chord for «output», not op-amp, just in case; the output stage is said to consist of discrete components. In any event I trust Rob to not change anything to the worse.

 

[x RELIC x]

I honestly don't think Rob would change his philosophy on his OP (output stage) with Hugo 2. It's been one of his driving factors for transparency. All I see is more power output and less battery life, not a re-design of the core principle of the implementation that has been a large factor of Chord DACs' success.

 

[Deftone]

Hugo 2 is going to have a Poly like module. I'm calling it!  

Hugo 2 module £2,000?

 

[maxh22]

Hugo 2 module £2,000?

That would be quite an absurd price lol

 

[miketlse]

That would be quite an absurd price lol

The Hugo2 already has the functionality of poly, apart from the sd card slot.
It would seem a lot of effort to have a Hugo2 module for just the SD card slot, so what other goodies will be incorporated? 

 

[Arpiben]

What do you mean by «OP»? This acronym was used by Chord for «output», not op-amp, just in case; the output stage is said to consist of discrete components. In any event I trust Rob to not change anything to the worse.

Indeed I mixed up acronyms if not more...I should have written one single OP stage with a minimum of discrete components. Since distorsion has been improved I should not worry if more or better transistors have been used. I do trust Rob.
Thanks Jazz & Relic.

 

[ColdFlo]

Wow I just got my Mojo(I like the spanish pronunciation, definitely not a cha ord moyo(some people have no taste)) and everything I could think of as an advancement or next feature after studying this tech they had already finished and now the product is launching.  More in custom circuit fpga "tuning", more array elements, more taps, more resolution, which fpga is in this one does anyone know exactly?, more power, DSD NATIVE(this was one limiting thing previous), DSD512, 768khz sampling, man this is one impressive company. The Mojo when I got it, my brain and ears heard picosecond snow sibilance that vaporized upon impact because my brain had never heard attack that fast and could not handle it..... and then in a matter of a couple of hours that all disappeared(i kind of miss it).  Someone should be knighted over such an invention.............  I was set on buying a Hugo this summer but now its going to be a Hugo 2.  Just wish I could get in custom circuit EQ shaping with god resolution god sampling output(oops I just gave away the features of the Hugo 3).
 
No the filters are the custom circuit in the FPGA from the limited amount I have read they voice them in the FPGA firmware not with the amp stage.  I dont know how a pulse array works to really discuss any of it in detail....(goes to read pulse array thread....)
 
Yes the voicing occurs in the dac not in the amp, which is after the pulse array.  I assume the dac feeds the pulse array and the voicing is done in the fpga dac custom circuit.

 

[Arnav Agharwal]

The sound signature filters would mean addition of an analog component in the output stage. Would it make any measurable difference to the amplification characteristics of the Hugo 2? Just curious.

 

[ChiAki]

Might I expect the filters to still function in DAC-only mode?
 
I'm planning to try out my amps and use the Hugo 2 as a reference DAC

 

[Arnav Agharwal]

  Might I expect the filters to still function in DAC-only mode?
 
I'm planning to try out my amps and use the Hugo 2 as a reference DAC

Assuming there is no major redesign of their core architecture, the Hugo 2 should not have a conventional amplification stage. The line-level output is simply the DAC output at 3V. This means that the DAC signal will always go through the filters, if applied.
 
For details about the architecture of their current output stage, you can check out the "Interview with Rob Watts" section in this post from the Mojo FAQ thread.

 

[x RELIC x]

.....

No the filters are the custom circuit in the FPGA from the limited amount I have read they voice them in the FPGA firmware not with the amp stage.  I dont know how a pulse array works to really discuss any of it in detail....(goes to read pulse array thread....)

Yes the voicing occurs in the dac not in the amp, which is after the pulse array.  I assume the dac feeds the pulse array and the voicing is done in the fpga dac custom circuit.

In simple terms, Pulse Array is the discrete DAC invented by Rob Watts. The FPGA is for the filters (WTA - Watts Transient Aligned filter) and other functions. The 'amp' has traditionally been just the critical I/V stage of the DAC (Current to Voltage) in Chord DACs, very much the line-out function for all outputs including headphone out. Essentially you are correct, no seperate amp in the path and voicing (or lack of) is from the DAC.

This post about Hugo 1 may clarify things:

http://www.head-fi.org/t/702787/chord-hugo/1830#post_10459450

Spoiler: Warning: Spoiler!

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. 

 

[Mython]

 

  Might I expect the filters to still function in DAC-only mode?
 
I'm planning to try out my amps and use the Hugo 2 as a reference DAC

Assuming there is no major redesign of their core architecture, the Hugo 2 should not have a conventional amplification stage. The line-level output is simply the DAC output at 3V. This means that the DAC signal will always go through the filters, if applied.
 
For details about the architecture of their current output stage, you can check out the "Interview with Rob Watts" section in this post from the Mojo FAQ thread.

 
 
x RELIC x posted relevant stuff about Rob's approach to filters (thanks), and here is some info about Rob's approach to output stages (much of this applies to both Mojo and Hugo 1, and probably quite a bit to Hugo 2, too, even though there have been some changes to Hugo 2s output stage vs Hugo 1):
 
 

Spoiler: Regarding Mojos output stage

 

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

 


 
 

Spoiler: Why is Mojos output Single-ended, and not Balanced?

 

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 

 
 

 

Spoiler: About Mojos Output-Impedance

 

@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

 

 

[Christer]

   
That's certainly not the case. «Frequency response: 20 Hz – 20 kHz  ± 0.2 dB» indicates the deviation from linearity within 20 Hz and 20 kHz (thus the audio band), not the disposable bandwidth, which will be much larger with hi-res recordings. I don't have measuring data for the classic Hugo at hand, but here are some for the TT.
 
 
I don't think that's true – it would be a regrettable turning away from the puristic original approach with no extra headphone amp for maximum transparency and accuracy. In fact both Hugos are said to have discrete class A output stages, so I'm confident that the design has stayed the same. See the slide show: «Hugo has a discrete class A OP stage integrated into the DAC output amplifier and filters» – which is the very same as in the original Hugo.

Thanks for your input again Jazz,

I  absolutely agree that maximum transparency  and accuracy are  most important  factors. It seems I misinterpreted the class A OP stage´s function.
But I sure hope they have made it  quite a bit more powerful than HUGO´s, because the lack of weight, power, body and authority on large scale classical with difficult to drive headphones is/was one of several  drawbacks of HUGO, not to mention how poorly it worked in at least my home system via speakers.
But it seems they have fixed those problems too. I noted that  now there is a remote control included as well. Hopefully it will be possible to adjust volume via the remote.Otherwise I see little reason to include one. Using HUGO direct to poweramp was a real pia having to get up and try to adjust volume on HUGO without it reacting with just loud hum after having  painstakingly taped it into a into workable position. I also count on them to have made much better sturdier analogue out RCA ports than on HUGO. I am keeping my fingers crossed that all in all, it will be so obviously audibly much better with my reference masterfiles than HUGO, that I can´t resist upgrading to HUGO 2.
If it proves to be the sonic marvel promised both as a portable and home DAC I will buy one ASAP.

 

[ChrisIsAwesome]

That's certainly not the case. «Frequency response: 20 Hz–20 kHz  ± 0.2 dB» indicates the deviation from linearity within 20 Hz and 20 kHz (thus the audio band), not the disposable bandwidth, which will be much larger with hi-res recordings. I don't have measuring data for the classic Hugo at hand, but here are some for the TT.

Thanks, i didnt realise.
Reading through how pulse array works im concerned that the hugo may not be a reference device but one tuned by rob watts listening tests and impressions? Through wta filter and noise shapers.
Would that be true or am i understanding things wrong.