[aangen]

I think the reason why people cried snake oil on the Pono is Joe six pack knows who Neil Young is and may have heard about his device. Joe six pack is totally unaware that the Mojo exists. They aren't writing about it in monster truck magazine, or Rolling Stone for that matter. I don't have to hear a Pono to know it sounds great, knowing who designed it. I am certain of this though, everyone who listened to a Pono and could not hear a difference would say the same thing about the more expensive Mojo. But those who could appreciate the quality of the Pono would appreciate the quality of the Mojo.

 

[uzi2]

  I think the reason why people cried snake oil on the Pono is Joe six pack knows who Neil Young is and may have heard about his device. Joe six pack is totally unaware that the Mojo exists. They aren't writing about it in monster truck magazine, or Rolling Stone for that matter. I don't have to hear a Pono to know it sounds great, knowing who designed it. I am certain of this though, everyone who listened to a Pono and could not hear a difference would say the same thing about the more expensive Mojo. But those who could appreciate the quality of the Pono would appreciate the quality of the Mojo.

...and also at the time he chose to release the most lo-fi album ever.

 

[masterpfa]

  I think the reason why people cried snake oil on the Pono is Joe six pack knows who Neil Young is and may have heard about his device. Joe six pack is totally unaware that the Mojo exists. They aren't writing about it in monster truck magazine, or Rolling Stone for that matter. I don't have to hear a Pono to know it sounds great, knowing who designed it. I am certain of this though, everyone who listened to a Pono and could not hear a difference would say the same thing about the more expensive Mojo. But those who could appreciate the quality of the Pono would appreciate the quality of the Mojo.

It's surprising how people who would not be described as audiophiles react when listening to the Mojo for the first time. I often offer friends or family the chance to listen to my Mojo but with their own earphones/HP's/IEM's.
 
The smile is reward enough. But I must admit these people would not be spending £400 on a DAC in the first place.
 
But I should add these same people probably have never used a DAC as an add on and have made do until now with their mobile phones.

 

[AndrewH13]

  It's surprising how people who would not be described as audiophiles react when listening to the Mojo for the first time. 
 
 

 
How you finding your Finder X1s with the Mojo? Are they available to buy yet in the UK?

 

[LouisArmstrong]

Folks. Let's rock. It's Mojo time!

 

[masterpfa]

   
How you finding your Finder X1s with the Mojo? Are they available to buy yet in the UK?

These are my favourite combination. I use the Finder on most occasions as they are quick and easy to inset and remove. For me great sound too.

They are only available currently online as far as I am aware but Echobox will be one of the sponsors of CanJam London, may bring some along to sell?

 

[landroni]

The fact that you preserve the original samples as a precondition obviously doesn't make your interpolated samples any more accurate than any other method.

Preserving the original sample points when upsampling, in a non-cheating workflow, would imply having a perfect brickwall filter of infinite length as the interpolation filter. Because otherwise there's simply no way the resampled audio can naturally pass through the original sample points.
 
IF (and that's a big IF) the megaburrito filter actually does what it claims, short of pulling an infinite computing miracle [...]
 

 
My understanding of the math involved is admittedly limited, however the DS approach almost guarantees that NONE of the original samples make it to the output of the DAC. This is by design, as one cannot squeeze e.g. a 16-bit sample through a 1-bit switch (even when you have 3-5 of them working in unison). While R2R D/A conversion works on levels (high-bit, low-speed information), DS D/A conversion relies on changes (at heart: 1-bit, extremely-high-speed information). This means that DS takes the samples, transforms them into changes suitable for the 1-bit switches, and then proceeds with the complex wizardry and black-box algorithms to get some sound out of the limited information retained... Re-creating the samples will generally only be going to approximate the original input; a very, very good approximation, but an approximation nonetheless, if we're actually interested in fidelity.
 
R2R doesn't have such limitations. If you take NOS designs, you can natively play ALL of the original samples (no conversions, no approximations, nothing). Of course, 16-bit 44.1 kHz sampling data will exhibit some (limited) aliasing distortion in the audible band, but not so 96 kHz material: all aliasing distortions will be constrained above 24 kHz frequencies, well outside the audible band. Schiit indeed claims that they can take original 44.1 kHz material and upsample it exactly while preserving all of the original samples. Personally I have no means to test these claims, but I am given to understand that Mike Moffat is preparing a white paper on these issues (which probably will be published when it will be published). The scant information that Mike has released on their algorithms is that to avoid the Parks-McClellan approximations a LOT of original (and unpublished) math work was required, including working around a division by zero problem. This may or may not address your "infinite computing miracle" concerns.
 
As for the claims of "timing accuracy" (i.e. time-domain optimization and associated 3D holographic, soundstage/imaging arguments), detecting a difference may depend heavily on the actual system you're using. It seems that accurate soundstage---Schiit MB, Theta, Mojo---,  becomes obvious in good speaker setups (less so in headphones). Not necessarily expensive, as even things like KEF LS50 or Elac B6/UB5 will likely do the job. So if you have a chance, try plugging a FiiO DAP into a speaker system and compare it with a Mojo or a Schiit MB device... Maybe this would help.

 

[Rob Watts]

  Question for Rob : some companies use electrolytic capacitors in their signal paths for their Dacs, but I recall reading you do not.
 
Can you kindly write a few words why you do no use them, and what effect they have?
 
The Hugo and Mojo sounds very fast and punchy to me, without these techniques !

Mojo/Hugo has one tantalum capacitor connecting the pulse array DAC to the combined output stage and I to V converter. Electrolytic capacitors have a lot more LF distortion than tantalum, and they sound soft and fat, and give some bloom to the lower registers. They can be used in a superficial attempt to balance the sound from a hard sounding unit; but like all balancing (or tuning) techniques, they ultimately do not work; you really need to solve the hardness problem in the first place. Some people like a soft bloated bass; but the problem with this is that adding softness means all bass sounds soft all the time. So when a musician/recording engineer/producer has a big soft bass, I want to hear it; but when they want a sharp fast tight bass, I want to hear that too; I don't want to hear the DAC adding its sound.
 
The other problem with electrolytic caps is the leakage current takes 3 months to stabilise; you get the lowest distortion and impedance after 3 months. My feeling is that this is the biggest reason for electronics break-in problem in that when I design devices that do not have electrolytic caps in critical areas, there is little or no actual hardware break-in, and warm-up period is very fast.
 
There are other factors about warm up and break-in, but they are more DAC specific problems, and these are issues I solved a very long time ago.
 
Rob

 

[AndrewH13]

  These are my favourite combination. I use the Finder on most occasions as they are quick and easy to inset and remove. For me great sound too.

They are only available currently online as far as I am aware but Echobox will be one of the sponsors of CanJam London, may bring some along to sell?

 
Will catch them and have a listen, thanks. Sounds like ie800s which I like with Mojo, one dynamic driver, extended freqs and super light.

 

[HIFIMAD]

Hi i have had a 2 hour demo of all chord dac s and the mojo is the best match for my headphones grado ps1000e the mojo will be a massive upgrade for you .its warmer sounding than the other chord  dac s and will be a good match with all headphones. i would like to chord to make a mojo t t .
 
Hi no its much warmer more bass hugo is bright more detailed
 
no you have to buy a camera connection kit for it to work on i devices

 

[Rob Watts]

   
My understanding of the math involved is admittedly limited, however the DS approach almost guarantees that NONE of the original samples make it to the output of the DAC. This is by design, as one cannot squeeze e.g. a 16-bit sample through a 1-bit switch (even when you have 3-5 of them working in unison). While R2R D/A conversion works on levels (high-bit, low-speed information), DS D/A conversion relies on changes (at heart: 1-bit, extremely-high-speed information). This means that DS takes the samples, transforms them into changes suitable for the 1-bit switches, and then proceeds with the complex wizardry and black-box algorithms to get some sound out of the limited information retained... Re-creating the samples will generally only be going to approximate the original input; a very, very good approximation, but an approximation nonetheless, if we're actually interested in fidelity.
 
R2R doesn't have such limitations. If you take NOS designs, you can natively play ALL of the original samples (no conversions, no approximations, nothing). Of course, 16-bit 44.1 kHz sampling data will exhibit some (limited) aliasing distortion in the audible band, but not so 96 kHz material: all aliasing distortions will be constrained above 24 kHz frequencies, well outside the audible band. Schiit indeed claims that they can take original 44.1 kHz material and upsample it exactly while preserving all of the original samples. Personally I have no means to test these claims, but I am given to understand that Mike Moffat is preparing a white paper on these issues (which probably will be published when it will be published). The scant information that Mike has released on their algorithms is that to avoid the Parks-McClellan approximations a LOT of original (and unpublished) math work was required, including working around a division by zero problem. This may or may not address your "infinite computing miracle" concerns.
 
As for the claims of "timing accuracy" (i.e. time-domain optimization and associated 3D holographic, soundstage/imaging arguments), detecting a difference may depend heavily on the actual system you're using. It seems that accurate soundstage---Schiit MB, Theta, Mojo---,  becomes obvious in good speaker setups (less so in headphones). Not necessarily expensive, as even things like KEF LS50 or Elac B6/UB5 will likely do the job. So if you have a chance, try plugging a FiiO DAP into a speaker system and compare it with a Mojo or a Schiit MB device... Maybe this would help.

Except the job of a DAC is NOT to reproduce the sampled data perfectly but to reproduce the original bandwidth limited analogue signal that was in the ADC before the signal was sampled. And to do this one must convert from a sampled signal and convert it to a continuous waveform - and that actually implies infinite oversampling, something that a R2R DAC can't do as they are limited to 16FS oversampling due to speed and glitch problems. That's one reason (there are many others too) why Mojo filters to 2048FS and has its DAC run at 104 MHz, unlike any other non Chord DAC's.
 
The aliasing is certainly not limited - yes you can't hear the images directly - but what they do is degrade the timing of transients, which you categorically can hear and to extremely low levels. So aliasing makes a huge difference to sound quality, as it degrades the accuracy of transient information. Transients are used by the brain for perceiving sound-stage, pitch, timbre and of course the starting and stopping of notes. Ever wondered why conventional digital was so poor at reproducing timbre, why you can't follow the bass tune, why it does not image properly or why it all sounds so un-musical? Its mostly down to the uncertainty of transients caused by sampling the continuous analogue signal. Fortunately this is a DAC design problem - with an infinite tap length infinitely oversampled FIR filter it will perfectly recover the original bandwidth limited analogue signal in the ADC - its just that conventional DAC's do not do a good enough job of this.
 
Another major problem with R2R DAC's is their complete inability to accurately reproduce small signals, as it is impossible to perfectly match the resistors - this is categorically not a problem for my pulse array DAC's as element mismatch creates fixed noise not distortion, as all the elements carry the audio signal (unlike R2R DAC's).
 
Rob

 

[Mojo ideas]

I think the reason why people cried snake oil on the Pono is Joe six pack knows who Neil Young is and may have heard about his device. Joe six pack is totally unaware that the Mojo exists. They aren't writing about it in monster truck magazine, or Rolling Stone for that matter. I don't have to hear a Pono to know it sounds great, knowing who designed it. I am certain of this though, everyone who listened to a Pono and could not hear a difference would say the same thing about the more expensive Mojo. But those who could appreciate the quality of the Pono would appreciate the quality of the Mojo.

An interesting viewpoint but when we played Mojo to Graham Nash he said it was like being back in the studio with Neil and when asked would he like to have one he simply stated ***** Yes! So perhaps news of mojo will travel even in exulted circles.

 

[tkteo]

  The aliasing is certainly not limited - yes you can't hear the images directly - but what they do is degrade the timing of transients, which you categorically can hear and to extremely low levels. So aliasing makes a huge difference to sound quality, as it degrades the accuracy of transient information. Transients are used by the brain for perceiving sound-stage, pitch, timbre and of course the starting and stopping of notes. Ever wondered why conventional digital was so poor at reproducing timbre, why you can't follow the bass tune, why it does not image properly or why it all sounds so un-musical? Its mostly down to the uncertainty of transients caused by sampling the continuous analogue signal. Fortunately this is a DAC design problem - with an infinite tap length infinitely oversampled FIR filter it will perfectly recover the original bandwidth limited analogue signal in the ADC - its just that conventional DAC's do not do a good enough job of this.
 

Mr Watts,
 
am I correct in inferring that your design principles for WTA therefore seeks to approach the challenge of reproducing the original analog signal at a more fundamental level, so to speak, and therefore the question I posed about pre-ringing and post-ringing is really not the crux of the challenge. Something that conventional DACs have to address but yet does not resolve deeper issues about digital audio?

 

[Rob Watts]

  Mr Watts,
 
am I correct in inferring that your design principles for WTA therefore seeks to approach the challenge of reproducing the original analog signal at a more fundamental level, so to speak, and therefore the question I posed about pre-ringing and post-ringing is really not the crux of the challenge. Something that conventional DACs have to address but yet does not resolve deeper issues about digital audio?

For a interpolation, or more accurately reconstruction filter, if you use an infinite tap length sinc FIR filter, you will reconstruct the original analogue signal in the ADC completely perfectly with no change whatsoever. But such a filter will have an inifinite amount of post and pre ringing - and this contradiction is solved when you realise that an impulse is not a legal signal, as it contains the same energy at FS/2 as at 1 kHz - so it is not bandwidth limited which requires zero output at FS/2. The idea that pre ringing is audibly bad is mistaken as an illegal (from sampling theory) test signal is being used. The filters that have the maximum pre-ringing using a non bandwidth limited signal is actually much more accurate in that when using a proper bandwidth limited signal will give the least differenct to the original un-sampled signal.
 
So yes you are correct - I am trying to perfectly reproduce the original un-sampled analogue signal, and the best filter is one that has an infinite amount of pre-ringing using an illegal non bandwidth limited impulse response. The WTA filter is much closer to an ideal sinc function, so has huge levels of pre-ringing - but using a proper bandwidth limited signal will return a signal that is closer to being unchanged than any other reconstruction filter available today.
 
And that's the primary reason why Mojo sounds like "you are there" because it more accurately reconstructs the analogue signal before it was sampled.
 
Rob

 

[tuna47]

I am testing the 800s soon and wonder if the mojo will drive them to their max