[zolom]

My Mojo connected to the Arrow 5Tx amp, to enhance bass. Great results.

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[jmills8]

My Mojo connected to the Arrow 5Tx amp, to enhance bass. Great results.

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[ThomasHK]

My Mojo connected to the Arrow 5Tx amp, to enhance bass. Great results.
[attach]1757453[/attach]


[attach]1757454[/attach][/quote]

Enhancing bass on SE846? Spoken like a true basshead! :D

 

[ThomasHK]

This thread moves too fast - every time I come back to it there's been more than 100 posts, so sorry if this has already been answered

Actually Rob Watts himself put forward the most convincing theory as to why USB cables can sound different. However it's way back in this thread and there's no way I'm spending the next day looking for it.

In a nutshell it comes down to how susceptible the USB cable is to interference.

The interference doesn't affect the digital signal, it's too low for that, so the bits come through verbatim - there is no change to the data. However in a DAC/AMP that isn't galvanically isolated that interference propagates through the DAC/AMP and effectively raises the noise floor of the analog parts. This isn't audible on it's own - dead silent parts will still sound dead silent to our ears. However it will have an effect on the analog signal. A cable that highly susceptible to interference may well make the highs a bit more grainy, which is often interpreted by our ears as a slightly brighter more detailed treble. This would account for why people say this cable sounds dull or this cable sounds clear. The truth is that the duller cable is more likely to be more effective in terms of interference rejection, which means, from an engineering perspective, that it's the better cable, despite our ears preferring the noisier inferior cable.

The choice of conductor is actually pretty irrelevant over the lengths of USB cable - you could make a cable with a steel conductor and there wouldn't be any difference from that of a copper or silver cable. It's how susceptible the cable is to interference that makes the difference.

If you want to hear the Mojo how it should sound then disconnect both the signal and charging cable, and connect it via the optical input, as this effectively isolates the Mojo and minimises interference. That said you may be disappointed as the optical tends to sound softer compared to a USB cable.

So, no fairies or pixie dust - just a pretty solid engineering explanation.

I was following your train of thought up to the point where you said that noise in the signal would make the system sound brighter or grainy. Surely whatever noise in the system that is not signal related (as you point out) will be so many dB's below the signal level and hence undetectable.

 

[x RELIC x]

I was following your train of thought up to the point where you said that noise in the signal would make the system sound brighter or grainy. Surely whatever noise in the system that is not signal related (as you point out) will be so many dB's below the signal level and hence undetectable.

According to the designer of the Mojo that's exactly what happens. His very extensive and controlled listening tests have drawn him to this conclusion (based on what's he's posted in the threads). He also has been quite surprised by our brains ability to resolve very fine subtleties in noise floor modulation and this is a large motivator for his designs. Also, from his observations, RF noise is like a fungus and difficult to get rid of. His desire to eliminate RF noise in the audio path (including entering the DAC) is indeed to reduce hardness in the sound reproduction.

He's commented quite a bit about this and you can read up on it in the third post of this thread and elsewhere on Head Fi. If you're really interested you can just read the All Posts from Rob Watts' profile, including his epic post on listening tests in the link below.

HIGHLY RECOMMENDED READING if you want to know the Mojo designers thoughts on listening tests and what we perceive:

http://www.head-fi.org/t/800264/watts-up#post_12457933

Spoiler: Rob on listening tests

I thought my first blog post should be non technical, and frankly the only non technical audio related subject I could think of that people may find interesting was listening tests - but I guess this is pretty fundamental subject for audio. After all, it's the main thing that separates the extreme objectivists (you don't need to listen it's all nonsense) from the extreme subjectivists (you don't need to measure it's all nonsense) argument. Its at the heart of the major discourse on Head-Fi - a poster says product xyz sounds great,another politely states your talking twaddle - of course they are (hopefully) arguing on the sound quality based upon their own listening tests, biases and preferences. Indeed, I often read a post about a device I know well, and can't relate the posters comments with what I think about the device. Sometimes this is simply different tastes, and I can't and won't argue with that - whatever let's you as an individual enjoy music is perfect for you, and if its different for me then that's fine - vive la différence. But sometimes the poster simply gets it wrong, because they do not have the mental tools to accurately assess sound quality. Over the many years I have developed listening tests that tries to objectively and accurately assess sound quality. These tests are by no means perfect, and I admit that listening tests are very hard and extremely easy to get wrong - that's why it's important to try to get more accurate results, as its very easy to go down the wrong path.

Another problem is sensitivity - some people hear massive changes, some can barely discriminate anything at all. Fortunately, I consider myself in the former camp, but I don't know how much is innate or through training (I have done lots of tests in my time...) Certainly, having an objective methodology does help, even if it's only about being able to more accurately describe things. 

I also would like to talk about how listening tests are actually used in my design methodology, and what it is I am trying to accomplish when designing.

Assumptions, Assumptions...  

We all make assumptions; otherwise we couldn't get on with life, let alone get anything done. But the key for designing is to test and evaluate one's assumptions, and verify that the assumptions make sense. The key is about thinking where the assumptions lie and then evaluating whether the assumption is valid. For example, I am assuming you are an audiophile or somebody that is interested in audio. Valid? Yes, otherwise why would you be on Head-Fi. I am also assuming you know who Rob Watts is. Valid? Not really, you may be new to Head-Fi or know nothing about Chord. So quick summary - I am an audiophile, a designer of analogue electronics (started designing my own gear in 1980) then in 1989 started designing DAC's. Frustrated by the performance of DAC chips, in 1994 I acquired digital design skills and started designing my own DAC's creating pulse array DAC technology using a new device called FPGA's. These devices allowed one to make your own digital design by programming an FPGA. I then became an independent design consultant, and started working with Chord, and the first product was the DAC 64. This was unique, in that it was the first long tap length FIR filter (the WTA filter). In 2001 I started working with silicon companies, and designed a number of silicon chips for audio. Most of my activity was in creating IP and patenting it, then selling the patents. Today, I only work on high end audio, having stopped working with silicon last year.

From my beginnings as an audiophile, I was intrigued about the physiology of hearing and spent a lot of time reading up about how hearing works. In particular, I was intrigued about how hearing as a sensory perception is constructed - we take our hearing for granted, but there is some amazing processing going on. 

The invention of the WTA filter with the DAC 64 nicely exposes the conventional engineering assumption - that the length of an FIR filter (an FIR filter is used in DAC's to convert the sampled data back into a continuous signal) does not matter, that all we need is a suitable frequency response. But if one looks into the maths of sampling theory, then it is clear that to perfectly recover the bandwidth limited signal in the ADC then an infinite tap length FIR filter is needed. It is also obvious to me that if you had a small tap length filter, then the errors would present themselves as an error in the timing of transients. Now a transient is when the signal suddenly changes, and from my physiology of hearing studies transients are a vital perception cue, being involved in lateral sound-stage positioning, timbre, pitch perception and clearly with the perception of the starting and stopping of notes. So how are we to evaluate the changes in perception with tap length? Only by carefully structured listening tests.

Another area where there are assumptions being made is designing using psycho-acoustic thresholds. The rational for this is simple. From studies using sine waves, we know what the human ear can detect in terms of the limits of hearing perception. So if we make any distortion or error smaller than the ear's ability to resolve this (hear it) then it is pointless in making it any better, as the ear can't detect it. On the face of it, this seems perfectly reasonable and sensible, and is the way that most products are designed. Do you see the assumption behind this?

The assumption is that the brain is working at the same resolution as our ears - but science has no understanding of how the brain decodes and processes the data from our ears. Hearing is not about how well our ear's work, but is much more about the brain processing the data. What the brain manages to achieve is remarkable and we take it for granted. My son is learning to play the guitar, and every so often the school orchestra gives a concert. He was playing the guitar, along with some violins, piano, and a glockenspiel. We were in a small hall; the piano was 30 feet away, violins and guitar 35 feet, glockenspiel 40 feet. Shut my eyes and you perceive the instruments as separate entities, with extremely accurate placement - I guess the depth resolution is about the order of a foot. How does the brain separate individual sounds out? How does it calculate placement to such levels of accuracy? Psycho-acoustics does not have a depth of image test; it does not have a separation of instruments test; and science has no understanding of how this processing is done. So we are existing with enormous levels of ignorance, thus it is dangerous to assume that the brain merely works at the same resolution as the ears.    

I like to think of the resolution problem as the 16 bit 44.1k standard - the ear performance is pretty much the same as CD - 96 dB dynamic range, similar bandwidth. But with CD you can encode information that is much smaller than the 16 bit quantised level. Take a look at this FFT where we have a -144 dB signal encoded with 16 bit:




So here we have a -144 dB signal with 16 bit data - the signal is 256 times smaller than the 16 bit resolution. So even though each quantised level is only at -96 dB, using an FFT it's possible to see the -144 dB signal. Now the brain probably uses correlation routines to separate sounds out - and the thing about correlation routines is that one can resolve signals that are well below the resolution of the system. So it is possible that small errors - for which the ears can't resolve on its own - become much more important when they interfere with the brains processing of the ear data. This is my explanation for why I have often reliably heard errors that are well below the threshold of hearing but nonetheless become audibly significant - because these errors interfere with the brains processing of ear data - a process of which science is ignorant off.

Of course, the idea that immeasurably small things can have a difference to sound quality won't be news to the majority of Head-fiers - you only need to listen to the big changes that interconnect cables can make to realize that. But given that listening tests are needed, that does not mean that objectivists are wrong about the problems of listening tests.

Difficulties in listening  

Placebo - convince yourself that your system sounds a lot better - and invariably it will. So your frame of mind is very important, so it's essential that when doing listening tests you are a neutral observer, with no expectations. This is not as easy as it sounds, but practice and forcing your mental and emotional state to be neutral helps.

Minimize variables. When lots of things change, then it becomes more difficult to make accurate assessments. So when I do a specific listening test I try to make sure only one variable is being changed.

Don't listen to your wallet. Many people expect a more expensive product to be naturally better - ignore it - the correlation between price and performance is tenuous.

Don't listen to the brand. Just because it is a brand with a cult following means nothing.  Ignore what it looks like too.

Do blind listening tests. If you are either unsure about your assessment, or want confirmation then do a single blind listening test where the other listener is told to listen to A or B. Don't leak expectation, or ask for value judgements - just ask them to describe the sound without them knowing what is A or B.

Remember your abilities change. Being tired makes a big difference to accuracy and sensitivity - more than 4 hours of structured AB listening tests means I lose the desire to live. Listening in unusual circumstances reduces sensitivity by an order of magnitude - double blind testing, where stress is put on listeners can degrade sensitivity by two orders of magnitude. Be careful about making judgements at shows for example - you may get very different results listening in your own home alone. Having a cold can make surprising differences - and migraines a few days earlier can radically change your perception of sound.  

Be aware - evaluating sound quality is not easy, and its easy to fall into a trap of tunnel vision of maximizing performance in one area, and ignoring degradations in other areas. Also, its easy to get confused by distortion - noise floor modulation, can give false impressions of more detail resolution. A bright sound can easily be confused with more details - distortion can add artificial bloom and weight to the sound. Its easy to think you are hearing better sound as it sounds more "impressive" but a sound that actually degrades the ability to enjoy music. Remember - your lizard brain - the part that performs the subconscious processing of sound, the parts that enjoy music emotionally - that can't be fooled by an "impressive" sound quality. Listen to your lizard brain - I will be telling how shortly.

Don't be afraid of hearing no reliable difference at all. Indeed, my listening tests are at their best when I can hear no change when adjusting a variable - it means I have hit the bottom of the barrel in terms of perception of a particular distortion or error, and this is actually what I want to accomplish.

Don't listen with gross errors. This is perhaps only appropriate for a design process - but it is pointless doing listening tests when there are measurable problems. My rule of thumb is if I can measure it, and it is signal dependent error, then its audible. You must get the design functioning correctly and fully tested before doing listening tests. 

Although I have emphasised the down side to listening, I find it remarkably easy to hear big changes from very small things - the ear brain is amazingly sensitive system. I once had an issue with a company accepting that these things made a difference, so I conducted a listening test with two "perfect" noise shapers - one at 180 dB performance, one at 200 dB performance. An non audiophile engineer was in the listening test, and afterwards he said that what really surprised him was not that he could hear a difference between two "perfect" noise shapers - but how easy it was to hear the differences.  

AB listening tests

Now to the meat of this blog, the actual listening tests. Here you listen for a set piece of music, and listen for 20 to 30 seconds, then go back and forth until you can assess the actual performance. The factors to observe or measure are:

1. Instrument separation and focus.

With instrument separation you are observing how separate the instruments sound. When this is poor, you get a loudest instrument phenomena: the loudest instrument constantly attracts your attention away from quieter instruments. When instrument separation gets better, then you can easily follow very quiet instruments in the presence of a number of much louder instruments. When instrument separation gets to be first rate then you start to notice individual instruments sounding much more powerful, tangible and real. Very few recordings and systems actually have a natural sense of instrument power - only very rarely do you get the illusion of a powerful instrument completely separate from the rest of the recording, in the way that un-amplified music can do. Poor instrument separation is often caused by inter-modulation distortion, particularly low frequency.  

2. Detail resolution.

Detail resolution is fairly obvious - you hear small details that you have not heard before- such as tiny background sounds or the ambient decay - and this is one measure of transparency. But its asymmetric - by this I mean you make an improvement, hear details you have not heard before, then go back, and yes you can just about make it out - once heard its easy to spot again with poorer detail resolution. Additionally, its possible to get the impression of more detail resolution through noise floor modulation - a brighter, etched sound quality can falsely give the appearance of more detail; indeed, it is a question of balance too; details should be naturally there, not suppressed or enhanced. This illustrates how difficult it is to get right. Small signal non-linearity is normally the culprit for poor detail resolution.   

3. Inner detail.

Inner detail is the detail you get that is closely associated with an instrument - its the sound of the bow itself on a violin for example, or the subtle details that go into how a key on a piano is played. Technically, its caused by small signal non linearity and poor time domain performance - improving the accuracy of transient timing improves inner detail.

4. Sound-stage depth.

A favorite of mine, as I am a depth freak. Last autumn we were on holiday in Northern Spain and we visited Montserrat monastery. At 1pm the Choir sing in the basilica, and we were fortunate enough to hear them. Sitting about 150 feet away, shutting ones eyes, and the impression of depth is amazing - and vastly better than any audio system. Why is the impression of depth so poor? Technically, small signal non-linearity upsets the impression of depth - but the amazing thing is that ridiculously small errors can destroy the brains ability to perceive depth. Indeed, I am of the opinion that any small signal inaccuracy, no matter how small, degrades the impression of depth.    

5. Sound-stage placement focus.

Fairly obvious - the more sharply focused the image the better. But - when sound-stage placement gets more accurately focused, the perception of width will shrink, as a blurred image creates an artificial impression of more width. Small signal non-linearity, transient timing and phase linearity contribute to this.    

6. Timbre.

This is where bright instruments simultaneously sound bright together with rich and dark instruments - the rich and smooth tones of a sax should be present with the bright and sharp sound of a trumpet. Normally, variation in timbre is suppressed, so everything tends to sound the same. Noise floor modulation is a factor - adding hardness, grain or brightness, and the accuracy of timing of transients makes a big difference.  

7. Starting and stopping of notes.

This is the ability to hear the starting of a note and its about the accuracy of transient timing. Any uncertainty in timing will soften edges, making it difficult to perceive the initial crack from a woodblock, or all the keys being played on a piano. Unfortunately, its possible to get confused by this, as a non linear timing error manifests itself as a softness to transients - because the brain can't make sense of the transient so hence does not perceive it - but in hard sounding systems, a softness to transients makes it sound overall more balanced, even though one is preferring a distortion. Of course, one has to make progress by solving the hardness problem and solve the timing problem so that one ends up with both a smooth sound but with sharp and fast transients - when the music needs it.      

8. Pitch and rhythm. 

Being able to follow the tune and easily follow the rhythm - in particular, listen to the bass, say a double bass. How easy is it to follow the tune? On rhythms its about how easy it is to hear it - but again, be careful, as it is possible to "enhance" rhythms - slew rate related noise modulation can do this.  In that case, things sound fast and tight all the time, even when they are supposed to be soft and slow.  

9. Refinement.

Clearly this links in with timbre, but here we are talking about overall refinement - things sounding smooth and natural, or hard and bright? Clearly, frequency response plays a major role with transducers, not so with electronics. Also, the addition of low frequency (LF) 2nd harmonic will give a false impression of a soft warm bass. Often I see designers balancing a fundamentally hard sound with the addition of LF second harmonic in an attempt to reduce the impact of the innate hardness - but this is the wrong approach, as then everything always sounds soft, even when its supposed to sound fast and sharp. In electronics, assuming THD is low, then noise floor modulation is a key driver into making things sound harder - negligible levels of noise floor modulation will brighten up the sound. Another very important aspect is dynamics and refinement - does the sound change as it gets louder - some very well regarded HP actually harden up as the volume increases - and harden up IMO in a totally unacceptable way.

"You know nothing Jon Snow"  

My favourite quote from Game of Thrones - but it illustrates the uncertainty we have with listening tests, particularly if done in isolation without solid measurements.

We are listening with recordings for which we do not know the original performance, the recording acoustic environment, nor do we know the equipment it was recorded with, the mastering chain, nor the source, DAC, amplifier, HP or loudspeaker performance in isolation. We are listening to a final result through lots of unknown unknowns. I can remember once hearing an original Beatles master tape played on the actual tape machine it was recorded with, using the actual headphones they used. It sounded absolutely awful. But then I was also lucky enough to hear Doug Sax mastering at the mastering labs - the equipment looked awful - corroding veroboard tracks on hand made gear - but it sounded stunning. So we are dealing with considerable uncertainty when doing listening tests. Its even more of a problem when designing products - how do you know that you are not merely optimizing to suit the sound of the rest of your system rather than making fundamental improvements to transparency? How can you be certain that a perceived softness in bass for example, is due to reduction in aberrations (more transparent) or increase in aberrations (less transparent).

Fortunately its possible to clarify or to be more sure with using two methods. First one is variation - all of the AB listening tests are really about variation - and the more variation we have, the more transparent the system is. So going back to the soft bass - if bass always sounds soft, then we are hearing a degradation. If it sounds softer, and more natural, but when the occasion allows sounds fast and tight too - then we have actually made an improvement in transparency. Again, its a question of being careful, and actually asking the question, is the system more variable. If it is more variable, its more transparent. So why bother with transparency and just make a nice sound? The reason I care so much about making progress to transparency is simply by listening to an un-amplified orchestra in a good acoustic makes one realize how bad reproduced audio is. Now I think I have made some big progress over the past few years - but there is still a way to go - particularly with depth, timbre variations and instrument power. This is why my focus is with the pro ADC project, as I will then be able to go from microphone to headphone/loudspeaker directly - my goal is being able to actually hear realistic depth perception exactly like real life.     

The second method of doing a sanity check on listening tests is with musicality...

Musicality

I ought to define what musicality actually is first, as people have different definitions. Some people think of it as things sounding pleasant or nice. That's not my idea of it - to me its about how emotional or involving the reproduced sound actually is. To me, this is the goal of audio systems - to enjoy music on. And plenty of people go on ad nauseam about this, so I am sorry to add to this. Merely talking about musicality does not mean you can actually deliver it - and it is something very personal.  

But it is important, and to test for musicality you observe how emotional and engaging the music actually is. The benefits of this approach is that your lizard brain that decodes and processes audio, and releases endorphins and makes your spine tingle, doesn't actually care whether you think it actually sounds better or not. And since enjoying music is what this hobby is about, then it is important to measure this. To do this, you can't do an AB test, you have to live with it, and record how emotionally engaging it actually is. That said, although its good as a test to check you are going in the right direction, its not effective for small changes, and it can only be based over many days or weeks with different music.

So listen to your lizard brain! I hope you got something useful from this.

Rob

You can take from this what you will.


Edit: Found in the Third post of this thread regarding RF and noise in the system (bold emphasis added by me).

Click below to expand:

Spoiler: Rob's comments on RF and how it changes the sound

Originally Posted by Rob Watts View Post

Originally Posted by musicheaven View Post

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

Originally Posted by Rob Watts View Post

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

 

[ThomasHK]

Thanks man, appreciate the links. The key word is modulation. Uncorrelated noise way down in level is imperceptible, noise modulated by the signal is not. It's the basis of how dither works.

 

[x RELIC x]

Thanks man, appreciate the links. The key word is modulation. Uncorrelated noise way down in level is imperceptible, noise modulated by the signal is not. It's the basis of how dither works.

I'm not sure, I'm not an EE nor have I been taught any established way of thinking on the matter. I only highlit Rob's observations in the spoilers that I added for his take on the subject and what he's observed from the effect of noise entering the system through USB cables. Of course you're welcome to make your own conclusions or assumptions about what we can actually hear (I mean this in the most respectful way).



Edit: This part of his listening tests post I linked is something I find quite funny actually, strictly regarding what's audible (not RF entering through USB):

"....Although I have emphasised the down side to listening, I find it remarkably easy to hear big changes from very small things - the ear brain is amazingly sensitive system. I once had an issue with a company accepting that these things made a difference, so I conducted a listening test with two "perfect" noise shapers - one at 180 dB performance, one at 200 dB performance. An non audiophile engineer was in the listening test, and afterwards he said that what really surprised him was not that he could hear a difference between two "perfect" noise shapers - but how easy it was to hear the differences.

....."

 

[Peter Hyatt]

Thanks man, appreciate the links. The key word is modulation. Uncorrelated noise way down in level is imperceptible, noise modulated by the signal is not. It's the basis of how dither works.

If in analog the particular sound wave is beneath our perception, does it impact another sound wave that is within our range, thus, impacting in some small way, our perception?

 

[Mython]

 

Thanks man, appreciate the links. The key word is modulation. Uncorrelated noise way down in level is imperceptible, noise modulated by the signal is not. It's the basis of how dither works.

If in analog the particular sound wave is beneath our perception, does it impact another sound wave that is within our range, thus, impacting in some small way, our perception?

 
 
This post, from the Hugo thread, is of interest, Peter:
 
 

Spoiler: Warning: Spoiler!

 

Does anyone know if Hugo or TT output bandwidth is limited (ultrasonic frequencies have been filtered)? I've read that a Naim power amp I'm considering does not like a full bandwidth input. Thanks.

No power amp or pre-amp "likes" a full bandwidth signal - not if that signal has out of band noise. Out of band noise (RF noise) makes the SQ brighter as it induces noise floor modulation - the noise pumps up and down with the wanted music signal. The brain detects this noise, and can't separate it from the wanted signal, and as noise sounds bright (hiss) it adds an edge to the perceived timbre of the instrument.
 
The technical reason why its important is down to the fact that audio electronics (whatever flavour) is non-linear at RF frequencies, and random RF noise inter-modulates with the wanted music signal, creating inter-modulation products that is within the audio bandwidth - hence noise that pumps up and down with the wanted signal. I have measured this problem countless times, so its a very real problem.
 
The brain is extremely sensitive to this issue - and when you remove noise floor modulation, it sounds a lot smoother and darker, with better instrument separation and focus. But it's easy to perceive the sound as less impressive; indeed I have seen many amp companies extolling the virtues of higher bandwidth - and its nothing to do with people hearing above 20 kHz - its just that if you do not bandwidth limit, you let in more RF noise, and this creates more noise floor modulation which then makes it sound brighter. When you hear noise floor modulation its easy to confuse it with better detail resolution, as it sounds brighter and etched - it's one reason why one has to be incredibly careful when doing listening tests, as its easy to perceive distortion as being better SQ. But of course, a bright and forced sound quality lacks musicality, and you will find it tires very easily - its one big source of listening fatigue.
 
So all of my DAC's are bandwidth limited to 50 or 60 kHz, as a DAC is a wonderful source of random RF noise. One of the very unusual things about my DAC's (from Mojo to Dave) is they do not have any measurable noise floor modulation, and I had to go to crazy lengths to do this from digital. But there is no point in having a DAC with no noise floor modulation if later on in the chain it has it due to the DAC's RF noise output, so hence the importance of bandwidth limiting.
 
A comment was raised about the HF filter with Dave; now that filter was designed to stop ADC noise shaper noise from DXD sources from creating noise floor modulation within Dave - not to stop that noise from corrupting the pre or power amp, as Dave's analogue filter will do this too.
 
So no need to worry about out of band noise.
 
Rob

 

 

[ThomasHK]

I'm not sure, I'm not an EE nor have I been taught any established way of thinking on the matter. I only highlit Rob's observations in the spoilers that I added for his take on the subject and what he's observed from the effect of noise entering the system through USB cables. Of course you're welcome to make your own conclusions or assumptions about what we can actually hear (I mean this in the most respectful way).



Edit: This part of his listening tests post I linked is something I find quite funny actually, strictly regarding what's audible (not RF entering through USB):

Don't worry man, I actually agree with Watts statements. I'm just pointing out that he's talking about noise that modulates with the audio signal. This is indeed perceptible and is the whole reason for dithering in digital audio. The dither added (i.e. Noise) is imperceptible if the bit depth is high enough.

Edit:I should point out that this noise/dither I'm talking about will be in every digital audio file you have. It's added during mastering when converting to different audio formats.

 

[Bengkia369]

To Chord:

I'm not asking too much, I just want a DAP module with Touchscreen for my Mojo in this Christmas!

 

[miketlse]

To Chord:

I'm not asking too much, I just want a DAP module with Touchscreen for my Mojo in this Christmas!

 
You asked the same question 2 days ago, shortly after @Mojo ideas had just provided the answer http://www.head-fi.org/t/784602/chord-mojo-the-official-thread-please-read-the-3rd-post/25875#post_13013837
 
I doubt the answer has changed in that short time. 

 
In the meantime, you can have many hours of fun and enjoyment over christmas, playing around with "what if" thoughts, trying to visualise what functionality this new product/module will provide.

 

[Bengkia369]

You asked the same question 2 days ago, shortly after @Mojo ideas
 had just provided the answer http://www.head-fi.org/t/784602/chord-mojo-the-official-thread-please-read-the-3rd-post/25875#post_13013837

I doubt the answer has changed in that short time. 

In the meantime, you can have many hours of fun and enjoyment over christmas, playing around with "what if" thoughts, trying to visualise what functionality this new product/module will provide.

He never directly answered my question.

 

[stuckonsound]

I've been searching this thread and found a couple comparisons to the Marantz HD-DAC1 as a DAC, but I'm curious if anyone has compared them as an all in one solution via the headphone output. I'm currently using the HD-DAC1 with HD700 and HE-500.

 

[Mython]

 

You asked the same question 2 days ago, shortly after @Mojo ideas
 had just provided the answer http://www.head-fi.org/t/784602/chord-mojo-the-official-thread-please-read-the-3rd-post/25875#post_13013837

I doubt the answer has changed in that short time. 

In the meantime, you can have many hours of fun and enjoyment over christmas, playing around with "what if" thoughts, trying to visualise what functionality this new product/module will provide.

He never directly answered my question.

 
 
LOL.
 
 
Well, try to put yourself in Chord's position, for a moment.
 
I am not speaking for them, but the way I see it:
 
 
Spend thousands of dollars taking a booth at one of the world's major electronics shows, to interact with the public and to make major product announcements.
 
 
Then see if you are eager to undermine that by prematurely spilling the beans to one impatient person on a forum...