Chord Mojo(1) DAC-amp ☆★►FAQ in 3rd post!◄★☆
Sep 15, 2016 at 11:37 PM Post #22,756 of 42,765
If your cable fails, no need to get ultra expensive optical cable. I use a cheap Livewire optical cable for 14.99 USD online for a 5 meter cable and it does 24/192 and DSD64 with zero issues whatsoever. Made in USA as well.

http://livewire-usa.com/eopt5/

Here's some pics with Mojo playing 24-bit 192 KHz




Here is Mojo again with DSD64 playback on optical:





I prefer USB now over optical since my usage with Mojo is strictly desktop and USB can upsample twice more than optical (384khz vs. 192khz) in windows sound properties. And also with my setup I hear a clear difference in timing of how music flows with Chord's native ASIO driver vs. Realtek driver that you are forced to use when running optical on Windows 10. I will keep your suggestion in mind if I were to use Mojo with mobile devices like AK in near future. Thank you for the recommendation. Happy listening! :)
 
Sep 15, 2016 at 11:45 PM Post #22,757 of 42,765
I use a USB to SPDIF converter such as the Wyred 4 Sound uLink in the pic above to just use one ASIO driver for all DACs and one USB port to output to 3 different DACs at the same time for quick A/B comparison. I did have the W4S uLink before I had the Mojo and Schiit Bifrost MB so might as well take advantage of it hehe.
 
Sep 15, 2016 at 11:48 PM Post #22,758 of 42,765
I prefer USB now over optical since my usage with Mojo is strictly desktop and USB can upsample twice more than optical (384khz vs. 192khz) in windows sound properties. And also with my setup I hear a clear difference in timing of how music flows with Chord's native ASIO driver vs. Realtek driver that you are forced to use when running optical on Windows 10. I will keep your suggestion in mind if I were to use Mojo with mobile devices like AK in near future. Thank you for the recommendation. Happy listening! :)


Of course you should determine what's most enjoyable for you, and only you can determine that. However, as Rob has mentioned, it's not the best idea to up-sample using the built in software of your PC as the Mojo is vastly superior at this and upsamples everything with MUCH more precision than the PC can do.



Quote:
Originally Posted by Rob Watts View Post

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:
Originally Posted by Rob Watts View Post

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:
Originally Posted by Rob Watts View Post

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.



This also pertains to the Mojo, but obviously not as sophisticated as the DAVE mentioned in the following quote.

Originally Posted by Rob Watts View Post

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


This information can also be found in the third post of this thread. :)
 
Sep 16, 2016 at 12:07 AM Post #22,759 of 42,765
Just to clarify, I am using Chords ASIO driver thats been downloaded from their website to upsample to 384khz 32bit in windows sound properties that I set to "Default device". Now, I am not sure if it's still windows that's doing the upsampling or Chord Mojo on the software side using their own ASIO driver that is doing the sampling. I think most likely the Chord ASIO since if I uninstall all audio softwares like Realtek, Chord ASIO in sound properties, I am left with "windows default sound" which reinstalls itself upon restart of the OS.

I find myself with downloaded mp3s and flac, the sound is best kept at 44.1 in default format of the file that has been downloaded. However, with streaming services like Tidal and Youtube and iTunes which in my only source, I find upsampling through "Chord's ASIO driver" to the highest rate gives me the best results. As always, IMO to my ears.
 
Sep 16, 2016 at 5:24 AM Post #22,760 of 42,765
I feel for Lavricable owners, I was a former owner of the cable as well a few months back. I simply couldn't take the inconsistent recognition of the iPhone after a while and it even had me contemplating if the Mojo is for me (after the apple CCK didn't work out either.) I never found rhyme or reason as to why it did what it did and this was far prior to iOS 10 with iPhone 5s.

I swallowed my pride and got refunded for the Lavricable cable after getting it "fixed" and charged two separate times. Got Penon and haven't looked back since.

No issues connecting, ever. It just works. Only thing I'd recommend is trying not to have the cable move around too much when connected to the mojo (which is a given in my opinion but life happens) or you run the risk of a loud static noise for a second or two. Very small and preventable problems compared to what used to be.
 
Sep 16, 2016 at 5:38 AM Post #22,761 of 42,765
Just to clarify, I am using Chords ASIO driver thats been downloaded from their website to upsample to 384khz 32bit in windows sound properties that I set to "Default device". Now, I am not sure if it's still windows that's doing the upsampling or Chord Mojo on the software side using their own ASIO driver that is doing the sampling. I think most likely the Chord ASIO since if I uninstall all audio softwares like Realtek, Chord ASIO in sound properties, I am left with "windows default sound" which reinstalls itself upon restart of the OS.

I find myself with downloaded mp3s and flac, the sound is best kept at 44.1 in default format of the file that has been downloaded. However, with streaming services like Tidal and Youtube and iTunes which in my only source, I find upsampling through "Chord's ASIO driver" to the highest rate gives me the best results. As always, IMO to my ears.


I think you'll find with more experimentation it varies on the source as to whether upsampling helps or hurts the sound quality. I tend to find upsampling streaming stuff that goes thru a DSP to be nice whereas with files I'm playing with a bit perfect player are best done at their actual sample rate. Of course, theres no reason to not do what sounds best to you.
 
Sep 16, 2016 at 6:00 AM Post #22,762 of 42,765
Regarding the Mojo's need for "bit-perfection" to shine, I've been under the impression that on the stock music app on iPhone doesn't play music bit-perfectly and needs something like Onkyo HF Player in order to achieve that.

Is this true or the only purpose behind this app is to play high res files on iPhone with Mojo? I use AAC for my phone and stream Tidal, plus I don't think any of my music is available in DSD.
 
Sep 16, 2016 at 6:10 AM Post #22,763 of 42,765
Bit perfect is way over rated. But if it makes you believe it sounds better than enjoy.
 
Sep 16, 2016 at 6:45 AM Post #22,764 of 42,765
Resampling is the way to go, 192 is much better than 44.1...your OS can do it...
 
popcorn.gif

 
Sep 16, 2016 at 7:07 AM Post #22,765 of 42,765
Thank you for the recommendation Relic. I am going to see if the Emotiva cable can do 192khz. If not, I am just going to bypass optical input all-together since I won't be able to get over not using Chord's ASIO driver with highest sampling rate available anyway. Hiwever, I am planning to buy a stock cable legnth USB cable from SysConcepts since it looks like they know what they are doing! We will see. :):D
It's very important that you make sure the cable is clicked firmly into place especially at the Mojo end as if it's not the cut end of the optical will not be aligned properly and transmission will be lost or only partially made.
 
Sep 16, 2016 at 7:16 AM Post #22,766 of 42,765
Resampling is the way to go, 192 is much better than 44.1...your OS can do it...

:popcorn:
. Sorry but that statement is not correct in the case of using Mojo. Mojos digital audio processing is far more capable than any PC or a laptop. Mojo has many tens of DSP cores running concurrently and will do a far better job of up sampling the Data internally so play the music in a none up sampled form for better results.
 
Sep 16, 2016 at 7:33 AM Post #22,767 of 42,765
Just to clarify, I am using Chords ASIO driver thats been downloaded from their website to upsample to 384khz 32bit in windows sound properties that I set to "Default device". Now, I am not sure if it's still windows that's doing the upsampling or Chord Mojo on the software side using their own ASIO driver that is doing the sampling. I think most likely the Chord ASIO since if I uninstall all audio softwares like Realtek, Chord ASIO in sound properties, I am left with "windows default sound" which reinstalls itself upon restart of the OS.

I find myself with downloaded mp3s and flac, the sound is best kept at 44.1 in default format of the file that has been downloaded. However, with streaming services like Tidal and Youtube and iTunes which in my only source, I find upsampling through "Chord's ASIO driver" to the highest rate gives me the best results. As always, IMO to my ears.


I'm pretty sure the up-sampling is still being done in Windows with the PC before it sends the bitstream signal to the Mojo. My preference is to not let the PC (Mac in my case) mangle the file before the Mojo can do it's own sophisticated processing. I believe the Windows Chord ASIO driver is simply a communication tool to ensure packets are sent over the USB correctly, but like I said earlier... if you enjoy it then that's what matters to you.
 
Sep 16, 2016 at 7:47 AM Post #22,768 of 42,765
. Sorry but that statement is not correct in the case of using Mojo. Mojos digital audio processing is far more capable than any PC or a laptop. Mojo has many tens of DSP cores running concurrently and will do a far better job of up sampling the Data internally so play the music in a none up sampled form for better results.

Sorry, John...I'm being too subtle in my sarcasm. The popcorn emoticon was a poor hint :). You're dead right, there's little if any common reason to upsample in the OS.
 
Sep 16, 2016 at 7:59 AM Post #22,769 of 42,765
Ok Got a question. I have been listening to mojo using j river on my pc (flac files ripped from my huge CD collection). I listen through Nighthawk and a 2.1 channel system (all yamaha gear)...I built my PC using an asrock motherboard with a toslink optical out. The motherboard had realtek acl1150 dac and realtek controller on the optical output too.
 
When I listen to mojo through USB I can hear that there is a fair amount of noise floor modulation (the sound is brighter, more metallic sounding treble, less musical, less engaging). Optical obviously sounds smoother, warmer more engaging and more natural..
 
So that difference can be explained through noise floor modulation via USB due to no galvanic isolation on mojo
 
My understanding from reading Robs posts is that the spdif inputs use a buffer and are retimed in the FPGA so asynchronous USB transmission shouldn't make a difference since all data from all inputs is retimed anyway
 
Soooo what really has me scratching my head is why on earth does all my music sound soooo much better when played straight from CD using my yamaha CDS300 and audioquest forest optical straight in to mojo???? the sound is more musical, more engaging, vocals seam to project into the room more and there seams to be 3 dimensional aspect to each sound.....it's also warmer and more natural. The only time I have heard better sound was when I auditioned DAVE.
 
So if Flac is lossless and mojo is imune to jitter (cos all data is retimed in the FPGA), then why does mojo sound its best out of my CDS300????? I should also mention that I have eq disabled in J river to get bit perfect output and of course I am not up sampling - cos I want mojo to get the original untouched data
 
Sep 16, 2016 at 8:04 AM Post #22,770 of 42,765
  Sorry, John...I'm being too subtle in my sarcasm. The popcorn emoticon was a poor hint :). You're dead right, there's little if any common reason to upsample in the OS.

A bit like this?
blink.gif


 

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