Interesting articles

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[Antoine]

Not really articles but forum posts by engineers that design DAC's that give us a bit of an insight to what are some of the issues (like RFI noise, bad clocks, and dirty ground planes) in computer audio and DAC's and the way they try to tackle these issues in their designs. These are taken out of their contexts here a bit but are IMO still interesting in a general sense.

Of course there are more ways than one to tackle these issues, one may be better at addressing an issue than another but at least these posts show the care required in a design. Just gluing the chips together on a circuit board won't do. Workaround solutions like the many "USB optimizers" out there and even the Mutec working/having an effect IMO shows the DAC is not immune or properly/sufficiently addressing the issues.

I don't own products of either company, no other ties as well. Also not saying they're right, some of the things they write contradict with my own experiences.



Rob Watts, Chord Electronics
http://www.head-fi.org/t/766517/chord-el...t_12577889

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The reason why I am so confident that jitter is a non issue [in the Chord Dave] is because of a number of things:
 
1. USB operation gets its timing from the local Dave oscillator, and incoming data gets re-locked to the local clock.
 
2. When I add 2 uS (that's 2,000,000 pS of jitter) to the data input from the AP using optical or coax I measure absolutely no change whatsoever. Now that on its own is not enough, as I have had situations before where unmeasurable effects are audible - but not concerning jitter. I have always been able to hear an effect then measure it.
 
3. One way that an incoming data can effect the SQ is down to ground plane noise, and in the past this used to be a big issue, both in measurement and SQ. And it's technically possible that ground plane and power supply noise can affect the SQ (I have seen this many times before). But in the case of my modern DAC's I have been able to eliminate this issue by a combination of local RF filtering on power supplies, double layer ground planing,use of efficient local SMPS, and power efficient FPGA's, plus careful layout. Now this issue used to be a nightmare, particularly with the FPGA, when my DSP cores used power hungry FPGA fabric. I would have to construct a DSP core by creating my own multipliers (today I use dedicated FPFA resources that are extremely power efficient), and every time a new place and route occurred, I would get different sound and different measurements. Today this situation never happens for lots of reasons - better design of the ground planes, better local RF filtering, better quality of RF filters, and dramatically lower signal induced noise (actually this is thousands of times lower than ten years ago) from the FPGA. Today, different place and routes show no SQ changes, or measured changes. What I am alluding to here is that the noise from a jittery source can't upset the sound quality through induced noise, as it is now (as far as I can tell) completely isolated - its also one of the benefits of the USB galvanic isolation in that the USB processor gnd and PSU noise is isolated from Dave.
 
4. Pulse Array DAC is innately jitter insensitive. What is not readily appreciated is that different DAC architectures have very different sensitivity to clock jitter. DSD is horribly sensitive to jitter, R2R DAC's are very sensitive, but pulse array is innately insensitive. The reason for this is that signal switching activity is completely signal independent - it switches in exactly same way whether its reproducing 0 of fully positive or negative. Because of this, when I get some clock jitter, it only creates a fixed noise. Now one of the really cool things that happens today is that PC resources and simulation tools are so good today, in that I can write a simulation, and add some jitter to the simulation, then measure the results using an FFT. From this, I can see exactly what jitter and only jitter does - and this technique has revealed a few surprises. But what it has done is proven that adding random jitter creates zero signal correlated effects to pulse array - no distortion, noise floor modulation at all - just an insignificant level of unvarying random noise. This does not happen with other DAC architectures, as you will then get significant noise floor modulation, distortion and noise shaper related noise. This is because with the other types of DACs, the switching activity is signal related. So DSD has maximum switching reproducing zero, and no switching at 100% modulation. R2R has no activity for zero, but considerable switching activity when the signal changes.
 
As to RF noise yes it is like a fungal infection. In the mid 80's, when I began to appreciate the importance of RF noise, I created a RF noise mains filter, in order to eliminate the SQ changes that mains cable was making. This ended up being a scary design - a cascade of inductors and capacitors, with filtering from 100 kHz to over 1GHz. I even had to make my own PTFE air cored inductors to get the performance I needed. But it worked - you could absolutely not hear the effects of different mains cable before the filter, but I had to use insane levels of filtering.
 
So I know how crazily sensitive things are to RF noise, which is why I can't say for absolute certainty that RF noise from Dave or from the sources may or may not affect your system - simply as most power amplifiers are very RF sensitive. So it really is a case of YMWV. But I would like to make a few suggestions:
 
1. PSU design - as far as impedance is concerned, forget it in relation to RF noise. Now impedance is an important issue, but it has no bearing on RF noise. It can be important in that signal currents will cause distortion due to the OP impedance. That's why the reference power supply for the pulse array is less than 0.0005 ohms for each and every flip flop as that is an important source of distortion. But it has no bearing on RF noise, that is another issue. But today, filtering within the DAC can be done to a very high level of performance with modern SMD capacitors, inductors and ferrite beads. 
 
2. Source - my advice with Dave is to use a source that is convenient for you. So far, in my system - and again YMMV, I have failed to hear any significant difference between different sources.
 
3. RF noise in terms of SQ. When you reduce RF noise, things sound softer and warmer and smoother. Bass is rounder with less slam and impact. Now if it is a bit perfect input source you are making changes, the only thing that can make a difference to the SQ is RF noise, so go for the warmer and softer sound - as that is the most transparent - even if it sounds too smooth! If it is too soft, don't worry - it is not a problem - you just need to then improve transparency elsewhere in your system - such as better cables, changing where the loudspeakers are sited, different HP, EQ etc. One of the profound problems we have with audio is that making fundamental improvements may affect the balance of the system, so it is less optimum. The trick is to understand when you have indeed made a fundamental improvement but that gives you an unbalanced sound then make other changes in your system to restore the overall balance.
 
Happy listening, Rob
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George Klissarov, Exasound
http://www.computeraudiophile.com/f6-dac...post323528

I will try to explain our approach. There are three problems with computer-based high-end audio that affect sound fidelity:

1. Computer s are not good for producing accurate timing. Computer clock beats are irregular and jittery.
2. Computers generate huge noise caused by bad power supplies and digital switching.
3. Computer sound systems, including the standard drivers, are not created for high-end audio. They are created for multipurpose use and quality is sacrificed for versatility, compatibility and easiness of use.


There are two approaches for addressing these problems. The mainstream approach is to make an "Audiophile Grade Computer". The noise can be reduced by making audiophile grade computer power supplies and USB power cleaners. The accuracy of computer timing can be improved by removing parts of the operating system so there are fewer interrupt requests. Further reducing of CPU load can be accomplished by distributing the processing on two computers (like the Network Audio Adapter). Random experiments with exotic cables can provide some relief from USB transmitted noise and can improve USB timing.

In my opinion the "Audiophile Grade Computer" approach is about treating the symptoms, not the root cause of the problems. It is like using pain killers. There will be endless small improvements in this area, something to buy every three months.

exaSound approached these challenges from the opposite point of view. Since computers have these basic and inevitable issues, just because of the way they work, then the only real solution is to create immunity to these issues in the gear that is connected to them. The DAC hardware and software must be designed to solve the issues of computer jitter and noise, and to provide bit-perfect streaming. A properly designed DAC must be immune to computer originated timing issues, USB jitter and streaming errors. A properly designed DAC must be very insensitive to USBtransmitted noise. These requirements are basic. Digital errors, noise and jitter can be measured, and DAC manufacturers have to demonstrate how effectively these problems are solved before we even start talking about the aesthetic perception of sound fidelity, the hard to describe qualities of air, space, transparency and realism.  

exaSound DACs solve the problem of computer timing issues and USB jitter by using a large hardware-implemented FIFO buffer. For simplicity think about it as a memory chip that stores let's say 0.5 seconds of music. It is located in the DAC box, and our proprietary drivers and USB interface can fill it up with sound stream data 10 times faster than the playback rate. Therefore we can guarantee that there is always data in the buffer. It doesn't matter how bad the computer timing accuracy is, because it only affects the delivery of data to the buffer. The bits and bytes will sit in the buffer for about 0.5 second anyway. On the other end of the buffer is the I2S interface to the DAC chip. The timing of delivery of data from the buffer to the DAC chip is determined only by the quality of the master clock and the design of the surrounding circuits. This approach allows for timing precision that cannot be achieved with computers. Best of all it is predictable and consistent.

There are several key points to make here:

* I2S wires, the links between the hardware that reads from the buffer, the master clock and the DAC chip must be as short as possible, otherwise they become a source of jitter. I2S is not meant to be used with external wires. Having the clock as close to the DACas possible minimizes jitter. Having an atomic clock outside as a standalone fancy component is another old idea. The wire to the external clock causes jitter.
* Implementing the FIFO buffer with hardware, without using a CPU is a great advantage in terms of timing precision on the buffer output. Computer-based software-implemented buffer is a half-measure. It suffers from the same problems, the reading is performed by a CPU controlled by software and influenced by interrupt requests. The output data stream has to travel along wires to reach the DAC. Wires cause jitter.


To solve the problem of computer-originated common noise and ground-loops hum, exaSound DACs use ground isolation. Traditional USB filtering technologies have a common ground wire connecting the computer with the DAC. This wire acts like an antennae for the high-frequency noise. Ground isolation cuts this connection. The evidence is in the noise measurements published on our website. If the noise levels are extremely low and about the same with and without USB cable connection to the computer, then the isolation is working. If connecting an external USBfiltering device doesn't affect in a visible way the noise measurements, then the isolation is working.

Finally the third problem of computer-based high-end audio is the quality of the general purpose sound drivers. exaSound's proprietary drivers provide a number of advantages:

* Our drivers are designed for and work only in 32bit integer mode. This is the native format used by the Sabre ES9018 DAC chip. Our sound-streaming software and hardware is designed to utilise the full potential of this chip. This is one of the reasons why our DACs sound so much better than other Sabre based DACs.
* Our drivers guarantee bit-perfect operation with error correction.
* In our opinion our implementation of asynchronous USB operation is better than the one offered by the USB Audio 2.0 standard and Apple's Core Audio.


I hope I've managed to answer the question why USB cables don't deliver exciting improvements in sound quality when used with exaSound DACs. For the same reasons using an external USB to SP/DIF convertor is not a good idea. The external USB to SP/DIF device duplicates functionality that is already present inside our products. At the same time it has the disadvantage to have to downgrade its output for delivery via SP/DIF wires, instead of having a direct short I2S connection to the DAC chip.  

There are several devices inside every exaSound DAC - a USBpurifier, an asynchronous reclocking FIFO buffer, a fine DAC and a preamplifier with high-end volume control. You need to decide what is the best way for you to solve the problems of computer-based high-end audio. There is no need to pay for the same features twice - on the computer side and on the DAC side.  

The bottom line is that computers and DACs can produce spectacular sound. These days high-resolution recordings are produced with computers, and there is no reason why they may sound better when downgraded by recording on physical analogue media.


Best,

George
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