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Topic: (Not a) good explanation of jitter in TAS (Read 62959 times) previous topic - next topic
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(Not a) good explanation of jitter in TAS

Reply #100
Nope, first page, second paragraph:

Quote
A crystal derived, or externally provided high quality master clock is used to allow low jitter recovery of S/PDIF supplied master clocks.


The WM8804's separate clock is used to improve input signal recovery! Besides, a S/PDIF receiver usually needs some form of transmitter, else it would be a dead-end without anything to output into.

In contrast a Cirrus Logic CS8427 is a typical clock-less S/PDIF transceiver. The data sheet contains diagrams on page 58, which show, that its PLL  is only able to eliminate jitter above 10kHz without a separate clean clock.

(Not a) good explanation of jitter in TAS

Reply #101
Nope, first page, second paragraph:
Quote
A crystal derived, or externally provided high quality master clock is used to allow low jitter recovery of S/PDIF supplied master clocks.



Crystal oscillators are free-running, and operate at a frequency that is set by the crystal. Thei phase of a crystal oscillator is basically random. It is free-running. It does what comes naturally.

A SP/DIF receiver clock must have the *identical* (e.g. +/- nothing at all) frequency as the input signal. Its phase must also match that of the input signal.

How does a crystal oscillator come to have the identical same frequency and phase as the input signal?

A frequency-synthesized variable frequency oscillator that is based on a crystal generally will produce only frequencies that are related to the crystal's frequency by quotients of integers. Therefore, not *all* frequencies can be generated this way. If the integers are large, then a very large number of dfferent frequencies can be generated, but still not every possible frequency can be generated.

(Not a) good explanation of jitter in TAS

Reply #102
I'm not sure the metaphor is problematic. Metaphors are, after all, literary, not scientific. What is faulty is the assumption that lead to the metaphor, and upon which the metaphor stands: that jitter is as audible at anything close to significant levels. It's questionable whether it is audible at all once it gets past a modern, well-implemented DAC. Audible enough to be heard above the noise floor of an analog lover's reference system? Highly unlikely. But we will never see blind listening tests from that crowd. Only subjective claims.

Tim

(Not a) good explanation of jitter in TAS

Reply #103
30ns amplitude jitter with a 500hz jitter signal yields a sideband amplitude of.... -210db?

The sideband amplitude is a function of the frequency of the audio signal being subjected to jitter.  Sideband amplitude is not a function of jitter frequency (as suggested above).  The math is also incorrect.

If a 1 kHz tone is subjected to 30ns RMS sinusoidal jitter having a frequency of 500 Hz, the jitter-induced sidebands will have an amplitude of -77.5 dB relative to the amplitude of the 1 kHz tone.  This means we will have a sideband at 500 Hz (1kHz - 500 Hz) and at 1500 Hz (1 kHz +500 Hz).  Each sideband can easily reach levels that are above the threshold of hearing if the 1 kHz tone is played at reasonably loud levels.  The THD+N due to the sidebands is -74.5 dB (relative to the amplitude of the 1 kHz tone).  Audibility will be a function of masking curves.  Also please note that the distortion caused by jitter is not harmonically related to the audio an should be more audible than harmonic distortion.

If a 10 kHz tone is subjected to 30 ns RMS sinusoidal jitter having a frequency of 500 Hz, the jitter-induced sidebands will have an amplitude of  -57.5 dB relative to the amplitude of the 10 kHz tone.  The sidebands will occur at 9500 Hz and 10500 Hz.

The distortion due to jitter is given by:

20*log(2*PI()*AudioFrequency*JitterMagnitude)

where AudioFrequency is expressed in Hz
and JitterMagnitude is expressed in Sec

Sideband amplitudes will be 3 dB lower than the combined distortion number calculated with the above formula.
John Siau
Vice President
Benchmark Media Systems, Inc.

(Not a) good explanation of jitter in TAS

Reply #104
I'm a little bit confused by your math so I tried a simpler case.

Assume that the jitter is a 500 Hz sawtooth with peak-to-peak amplitude of 30 nSec.

Assume that you have a 1 kHz sine wave as your signal and that the jitter is synchronized in such a way that one cycle of the sine wave corresponds with the rising part of the sawtooth, and the next cycle corresponds to the falling part.

Now we have a slope in sample timing offset of +- 30 nSec / 1 mSec (amplitude divided by rise and fal times). This will have the effect of increasing or decreasing the frequency of each sine wave by 0.003%. We now have alternating 999.97 Hz and 1000.03 Hz sine waves.

So where do the signals at 500Hz and 1500Hz come from, and how would it be humanly possible to hear so small a frequency variation in the sine wave?

Sorry if I am way off base here, I am just trying to understand.

(Not a) good explanation of jitter in TAS

Reply #105
a Cirrus Logic CS8427 is a typical clock-less S/PDIF transceiver. The data sheet contains diagrams on page 58, which show, that its PLL  is only able to eliminate jitter above 10kHz without a separate clean clock.


Most AES/EBU or S/PDIF receivers have little or no jitter attenuation below 5 or 10 kHz.  The reason for this is that they must meet the AES jitter tolerance test that requires perfect data recovery in the presence of high-amplitude low-frequency jitter.  This data-recovery task demands a PLL with a corner frequency of at least 5 kHz.  Below 5 kHz the PLL tracks the jitter of the incoming signal.  Above 5 kHz the PLL begins to reject the jitter on the incoming signal.

The AES jitter tolerance specifications were based upon measurements of jitter levels on commercially available equipment.  Devices that meet the AES jitter tolerance specifications should be able to reliably recover data from nearly all source devices.  However, this does not imply that the clock that is recovered by the PLL is suitable for D/A or A/D conversion.

Direct use of the MCLK recovered from a S/PDIF (or AES) receiver will yield jitter-induced distortion products that are only 50 to 70 dB below the amplitude of the audio signal.

A second PLL is required to reduce the jitter on the clock that is recovered by the receiver chip.  This second PLL must control the frequency of a low-jitter oscillator.  It is common to use a VCXO (voltage controlled crystal oscillator) as the oscillator in this second PLL.  Other low-jitter oscillator designs are possible, but the VCXO makes the task somewhat easier.  The corner frequency of the second PLL should be much lower than the 5 kHz corner frequency of the first PLL.  1 Hz to 10 Hz corner frequencies are often used in high-quality professional audio equipment.  Remember there is no jitter attenuation below the corner frequency of the PLL.
John Siau
Vice President
Benchmark Media Systems, Inc.

(Not a) good explanation of jitter in TAS

Reply #106
30ns amplitude jitter with a 500hz jitter signal yields a sideband amplitude of.... -210db?

The sideband amplitude is a function of the frequency of the audio signal being subjected to jitter.  Sideband amplitude is not a function of jitter frequency (as suggested above).  The math is also incorrect.
You are correct; I was using FM math for the sidebands instead of PM. Thank you for the correction. (Methinks I need to read Dunn more often.)

Your math, unlike mine, also makes sense in light of existing listening tests.

So where do the signals at 500Hz and 1500Hz come from, and how would it be humanly possible to hear so small a frequency variation in the sine wave? Sorry if I am way off base here, I am just trying to understand.
You're confusing instantaneous frequency with physical frequency. Like I said, the best source for the basics of the math is probably:

http://www.nanophon.com/audio/jitter92.pdf

(Not a) good explanation of jitter in TAS

Reply #107
So where do the signals at 500Hz and 1500Hz come from, and how would it be humanly possible to hear so small a frequency variation in the sine wave?


Jitter phase-modulates the audio signal and produces upper and lower sidebands.  The jitter frequency determines how far the sidebands will be spaced away from the audio signal.  Low frequency jitter will produce closely spaced sidebands that tend to be well masked by the original audio signal.  High-frequency jitter produces sidebands that are widely spaced above and below the original signal and consequently are not as well masked.

Side band frequency is given by CarierFrequency +/- JitterFrequency when both are sinusoidal

Your example using the sawtooth is actually much more complicated mathematically.  The sawtooth jitter will generate many side bands not just the ones you identified.

John Siau
Vice President
Benchmark Media Systems, Inc.

(Not a) good explanation of jitter in TAS

Reply #108
The sideband amplitude is a function of the frequency of the audio signal being subjected to jitter.  Sideband amplitude is not a function of jitter frequency (as suggested above).


For small amounts of modulation, sine wave FM or PM  modulation of a sine wave will produce a pair of sidebands that are displaced from the carrier by the modulating frequency. Their amplitude is proportional to the strength of the modulation.

For large amounts of modulation, the sideband structure becomes complex with many sidebands. Their frequencies and the proportioning of energy among them is predicted by the Bessel functions.

Phase modulation and frequency modulation can be accomplished and analyzed by identical means. They are linked by differentiation or in the reverse, by integration of the modulating signal. The modulating signal is differentiated to produce phase modulation by means of a frequency modulator, and vice-versa.

Quote
If a 1 kHz tone is subjected to 30ns RMS sinusoidal jitter having a frequency of 500 Hz, the jitter-induced sidebands will have an amplitude of -77.5 dB relative to the amplitude of the 1 kHz tone.  This means we will have a sideband at 500 Hz (1kHz - 500 Hz) and at 1500 Hz (1 kHz +500 Hz).  Each sideband can easily reach levels that are above the threshold of hearing if the 1 kHz tone is played at reasonably loud levels.  The THD+N due to the sidebands is -74.5 dB (relative to the amplitude of the 1 kHz tone).  Audibility will be a function of masking curves.  Also please note that the distortion caused by jitter is not harmonically related to the audio an should be more audible than harmonic distortion.

If a 10 kHz tone is subjected to 30 ns RMS sinusoidal jitter having a frequency of 500 Hz, the jitter-induced sidebands will have an amplitude of  -57.5 dB relative to the amplitude of the 10 kHz tone.  The sidebands will occur at 9500 Hz and 10500 Hz.


The sidebands produced by a given amount of jitter are 20 dB larger for the 10x higher frequency carrier, because 30 ns is a 10x (+20dB)  larger fraction of the period of the 10 KHz frequency carrier.


Quote
The distortion due to jitter is given by:

20*log(2*PI()*AudioFrequency*JitterMagnitude)

where AudioFrequency is expressed in Hz
and JitterMagnitude is expressed in Sec

Sideband amplitudes will be 3 dB lower than the combined distortion number calculated with the above formula.


When you're looking at the magnitude of the spectrum produced by various distortion sources, you really can't tell from a magnitude-only plot  whether you're seeing AM or FM distortion. Often real-world distortion is a mixture.  Both AM and FM  produce sidebands that differ from the carrier by the modulating freqency. They both produce sidebands with equal magnitudes. However, the phase of the sidebands differs, which will cause you to observe unequal sidebands for a mixture of AM and FM of the same carrier by the same modulation.

(Not a) good explanation of jitter in TAS

Reply #109
When you're looking at the magnitude of the spectrum produced by various distortion sources, you really can't tell from a magnitude-only plot  whether you're seeing AM or FM distortion. Often real-world distortion is a mixture.


Very true!  Our own measurements of the D/A converters built into CD players show that all of the units tested suffered from both phase modulation (FM) and amplitude modulation (AM).  We traced these problems to ripple on the DC power supply rails, and to ripple in the ground system.  Some of the ripple was AC line related, but much of the ripple was caused by the servos that drive the read head and control the rotational speed of the disk.  This ripple was causing phase modulation of the oscillator (jitter), and amplitude modulation of the D/A reference voltage.  These problems were common to all of the CD and DVD players that we looked at in our lab.  The players ranged in price from $50 to $1200.  Our sampling of 4 or 5 players was very small, but it did indicate that AM and FM modulation problems may be very common in CD and DVD players.  The worst player produced sideband amplitudes that were only 50 to 60 dB below our recorded test tones.
John Siau
Vice President
Benchmark Media Systems, Inc.

(Not a) good explanation of jitter in TAS

Reply #110
There's no reason for a standard DAC to have a crystal. The device sending the digital data either provides a separate but parallel clock signal, or the clock is derived from the input digital data stream. The latter is far and away the most common situation.

The reason for adding a crystal is to achieve low-jitter performance at frequencies that exceed the cut-off frequency of the clock-recovery PLL.

The crystal is usually a VCXO (voltage controlled crystal oscillator) that is being controlled by a PLL.  The PLL includes a phase comparator and a low pass filter.  Above the cut off frequency of the low-pass filter, the jitter performance is determined by the stability of the oscillator.  Below the cut-off frequency of the low-pass filter,  the jitter performance is determined by the quality of the clock embedded in the digital input signal (AES or S/PDIF).  The crystal oscillator makes it much easier to achieve low-jitter performance above the PLL cut-off frequency.  The stability of the oscillator is especially important when the PLL cut-off frequency is very low (less than 100 Hz).  Low cut-off frequencies are required to eliminate low-frequency jitter, so this makes the use of a VCXO a great solution.  But the VCXO solution is not cheap.  To cut costs, the VCXO is often omitted, and the clock recovered by the digital audio receiver is simply wired directly to the D/A converter.  This is not good practice, but it is very common (and very inexpensive).
If you see a crystal in a DAC, either the DAC resamples asynchronously, or the crystal is actually there for the benefit of the ADC that is in the same box.

VCXOs and crystal oscillators look identical.  It is impossible to tell the difference without looking up data sheets on the oscillators.

If the crystal is fixed-frequency, this may be an indication that the DAC resamples asynchronously.  Some ASRC (asynchronous sample rate converter) ICs attenuate jitter, but most do not.  Most of the ASRC ICs we have tested have very little jitter attenuation below 5 kHz.  There are 3 or 4 ASRC ICs that have outstanding jitter attenuation that extends down to a few Hz and these few can outperform two-stage PLL solutions that employ VCXOs.
Asynchronous resampling has to violate the principle of bit-perfect data transmission.

Very true, but the quality of the ASRC is a function of how much DSP horsepower we are willing to expend.  The distortion artifacts of the better ASRC devices are below -140dB.  These distortion artifacts are as much as 100 dB lower than the jitter-induced sidebands produced by DACs that do not use a VCXO or a fixed-frequency crystal.
John Siau
Vice President
Benchmark Media Systems, Inc.

(Not a) good explanation of jitter in TAS

Reply #111
When you're looking at the magnitude of the spectrum produced by various distortion sources, you really can't tell from a magnitude-only plot  whether you're seeing AM or FM distortion. Often real-world distortion is a mixture.


Very true!  Our own measurements of the D/A converters built into CD players show that all of the units tested suffered from both phase modulation (FM) and amplitude modulation (AM).  We traced these problems to ripple on the DC power supply rails, and to ripple in the ground system.  Some of the ripple was AC line related, but much of the ripple was caused by the servos that drive the read head and control the rotational speed of the disk.  This ripple was causing phase modulation of the oscillator (jitter), and amplitude modulation of the D/A reference voltage.  These problems were common to all of the CD and DVD players that we looked at in our lab.  The players ranged in price from $50 to $1200.  Our sampling of 4 or 5 players was very small, but it did indicate that AM and FM modulation problems may be very common in CD and DVD players.  The worst player produced sideband amplitudes that were only 50 to 60 dB below our recorded test tones.


Interesting.  Have you seen Ian Dennis and Julian Dunn's white paper on possible causes of nonidentical CDP output from bit-identical CDs, from circa 1995?

www.prismsound.com/m_r_downloads/cdinvest.pdf

They posited servo-based effects as well.

(Found no solid evidence of audible effect though)

(Not a) good explanation of jitter in TAS

Reply #112
Interesting.  Have you seen Ian Dennis and Julian Dunn's white paper on possible causes of nonidentical CDP output from bit-identical CDs, from circa 1995?

www.prismsound.com/m_r_downloads/cdinvest.pdf

They posited servo-based effects as well.

(Found no solid evidence of audible effect though)

Yes, I am familiar with their work and I spent some time discussing it with Julian at an AES convention.  We had conducted our own testing several years later (in 2003) and saw similar problems with CD players.

We have a QSC ABX tester that we used to compare consumer CD players to prototypes of our DAC1 converter.  It was fairly easy to score perfectly on the ABX tests.  The CD players with modulation problems sounded like they had more midrange when played through their internal D/A converters than when played through the external DAC.  This was a rather surprising result given that both devices had nearly identical frequency responses.  The CD players tested and the DAC1 prototype both had very flat frequency response between 20 Hz and 20 kHz.  The frequency response of the DAC1 extended down to 0.1 Hz, but our playback system was limited to -3dB at 30 Hz.

Subsequent FFT analysis confirmed that the modulation-induced sidebands tended to fill the midrange of the audio spectrum.  Some of the added content in the midrange may have been due to high levels of IMD produced by the internal D/A converters and the output stages in the CD players.  Based upon our listening tests, we suspect that jitter is often perceived as a difference in frequency response.  Specifically, we suspect that jitter-induced sidebands fill in the midrange of the audio spectrum.  Much more investigation would be needed to determine audibility thresholds.  Nevertheless, we felt we had enough evidence to warrant developing a jitter-attenuation system.

The external prototype DAC was equipped with an Analog Devices AD1896 ASRC which rejects jitter above 1 Hz.  At 1 kHz the jitter attenuation of the AD1896 exceeds 100 dB.  Jitter tolerance tests and FFT analysis confirmed that the prototype DAC was essentially free from jitter-induced (FM) sidebands as well as AM sidebands.

I am a strong advocate of ABX testing, but I take a very conservative approach to product design.  I am not in the business of building perceptual encoders where the goal is just to reduce artifacts to a certain level of audibility (or just below audibility without wasting bandwidth by going too far).  Instead I have the luxury of building products that can reduce artifacts to a level that is low enough that I have no doubt that these artifacts are inaudible.  This is analogous to a "safety factor" built into a highway bridge.  A bridge that is designed to carry 100 Tons may be built with a 5:1 safety factor and may be able to carry about 500 Tons before failing.  This safety factor insures that the bridge will not fail under normal use.  Safety factors can be built into audio circuits at fairly low cost to insure that certain audio artifacts never reach audible levels.  It is often much easier and cheaper to over design than it is to create a design that is just good enough.  It is very hard to determine exactly how much jitter attenuation is needed to prevent audibility.  It is much easier to design in a generous safety factor.

Manufacturers of inexpensive consumer-grade audio equipment have a different goal:  How cheap can we make the product before most people will think it sounds bad?  No safety factor is needed because the average consumer will tolerate some audible artifacts.

I highly recommend Julian Dunn's papers on jitter.  They are required reading on this subject!
John Siau
Vice President
Benchmark Media Systems, Inc.

(Not a) good explanation of jitter in TAS

Reply #113
Quote
Much more investigation would be needed to determine audibility thresholds. Nevertheless, we felt we had enough evidence to warrant developing a jitter-attenuation system.


But this is exactly evidence that is in pressing need of being made public, IMO.  It would not be too surprising if artifacts in the midrange have a lower threshold , as it's where H. sapiens is most sensitive.  But I'd like to see such evidence.

I'm not meaning to berate you for focusing on the engineering/design end, but given the eternal subjectivist/objectivist debates, it frustrates me when people in the industry report private DBT results that show difference within 'hot button' categories -- e.g. digital formats, CDPs, cables, amps -- much less ones where 'it was fairly easy to score perfectly' --- but don't actually publish them or provide detail (I'm thinking too of Robert Stuart's brief anecdotal references to such blinds tests in his papers advocating hi-rez formats).  Your company makes fine DACs; one would think you'd be eager to publish solid proof that they differ audibly from mass-market stuff...that that overdesign actually pays off in more than just excellent specs. 

The extant literature on audibility of jitter is sparse and the results appear highly method-contingent.  The extant literature on CDP listening comparison is as tiny or tinier.  Both really could use some more systematic scientific investigation of audibility thresholds of measured CDP difference.

(Not a) good explanation of jitter in TAS

Reply #114
But if they published ABX tests, they would be marked in the HiFi community as people who doesn't hold some sort of "true values". I've never seen ABX test results from HiFi companies that make DACs, amplifiers, CD/DVD players, and so on. Maybe I am wrong about this, but I have the feeling that publishing ABX results would actually undermine their efforts in making (and eventually selling) good DAC.
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(Not a) good explanation of jitter in TAS

Reply #115
But if they published ABX tests, they would be marked in the HiFi community as people who doesn't hold some sort of "true values". I've never seen ABX test results from HiFi companies that make DACs, amplifiers, CD/DVD players, and so on.


Other than some really bad cheap stuff, and some relatively rare pathological cases, there's not a lot of differences to hear.

Quote
Maybe I am wrong about this, but I have the feeling that publishing ABX results would actually undermine their efforts in making (and eventually selling) good DAC.


The days of the DAC as a separate mainstream consumer audio component have been gone for some time.  DACs are now routinely encapsulated into power amplfiiers and receivers, which is where they IMO belong.

(Not a) good explanation of jitter in TAS

Reply #116
But if they published ABX tests, they would be marked in the HiFi community as people who doesn't hold some sort of "true values". I've never seen ABX test results from HiFi companies that make DACs, amplifiers, CD/DVD players, and so on. Maybe I am wrong about this, but I have the feeling that publishing ABX results would actually undermine their efforts in making (and eventually selling) good DAC.



Tag McLaren (amps) actually published one on the web some years ago, with the results supporting no audible difference between their gear and much more expensive gear.

I think they changed ownership after that, and the site has vanished.

(Not a) good explanation of jitter in TAS

Reply #117
I think there is a valid market for those in the luxury market who perceive the threshold of audibility as what is the "bare minimum". Enjoying the highest quality engineering available, regardless of minimum specifications of equivalence, is not wrong. I do not own a Benchmark product, but I have heard a DAC-1 once or twice, and am aware of the high reputation Benchmark holds, so I have no reason to doubt they belong in such a tier. I wouldn't mind having such a product, just to avoid even the barest inkling of misapprehension about my DAC - but probably also no small amount of conspicuous consumption.

However, it is a completely different thing to justify such overengineering on the basis of audibility below known thresholds, and there are of course shades of gray there with respect to if thresholds can be exceeded for test signals vs musical content etc. Like, there's nothing wrong about SACD in and of itself, but if you can't prove its superiority with a blind test (or nowdays perhaps several), there is something distinctly wrong about maintaining that claim in light of testing and considerable psychoacoustic justification.

So I find it something of a shame that many high-end audio firms cannot sell their products on their engineering alone, without needing to make claims about what will and won't be an audible improvement, which naturally introduces a conflict with academia. (I'm not talking about Benchmark specifically here - I know nothing about their literature.) Why can't luxury be for its own sake?

(Not a) good explanation of jitter in TAS

Reply #118
It was fairly easy to score perfectly on the ABX tests.  The CD players with modulation problems sounded like they had more midrange when played through their internal D/A converters than when played through the external DAC.


Interesting.

How did you match the playback levels of the DACs ?

(Not a) good explanation of jitter in TAS

Reply #119
It was fairly easy to score perfectly on the ABX tests.  The CD players with modulation problems sounded like they had more midrange when played through their internal D/A converters than when played through the external DAC.


Interesting.

How did you match the playback levels of the DACs ?


Time-synching is IME far more difficult than levelmatching.

(Not a) good explanation of jitter in TAS

Reply #120
Yes, for instant switching, time synchronisation can be a big problem.

But if the delay between the two dacs (internal and external) is small enough, the problem can be circumvented by muting the amplifier before any switch. In this case, a DC offset between the mass of the devices may introduce an audible click. So the muting must be done with the volume control. It must also be checked that the volume control completely mutes the audio. Some let pass a small signal. In this case it is always possible to mute the amplifier after having turned down the volume.

A switch is then done this way : Amplifier volume to zero / amplifier speakers off / ABX switch switched / amplifier speakers on / amplifier volume restored.

If two different CD players are compared, the synchronisation is more problematic. In this case, a switch consists in
Amplifier volume to zero / amplifier speakers off / both CD players stopped / ABX switch switched / both CD players launched and paused / both CD players manually unpaused at the same time / amplifier speakers on / amplifier volume restored.

(Not a) good explanation of jitter in TAS

Reply #121
Yes, for instant switching, time synchronisation can be a big problem.


To tell the whole story, if nearly instant switching is not available, then the potential sensitivity of the comparison is seriously, perhaps even debilitatingly compromised.

Quote
But if the delay between the two dacs (internal and external) is small enough, the problem can be circumvented by muting the amplifier before any switch.


Of course. The problem of transient-free switching was addressed in detail in Clark's 1978 JAES paper.  I actually worked out a general procedure for transient-free switching of both inputs and outputs in 1976-7, and provided it to Clark to include in his paper. The benchmark test was to do a switchover of both inputs and outputs of a integrated amplifier from RIAA input to speaker output. I know of no other switching problem that is not either like that problem or a subset of it.

Quote
If two different CD players are compared, the synchronisation is more problematic. In this case, a switch consists in
Amplifier volume to zero / amplifier speakers off / both CD players stopped / ABX switch switched / both CD players launched and paused / both CD players manually unpaused at the same time / amplifier speakers on / amplifier volume restored.


You've missed the point. The problem of synchronization is to have 2 CD players that are playing the identical same tracks at the same time offsets within a few dozen milliseconds or less over a useful amount of music for the purpose of comparison.

(Not a) good explanation of jitter in TAS

Reply #122
To tell the whole story, if nearly instant switching is not available, then the potential sensitivity of the comparison is seriously, perhaps even debilitatingly compromised.


A mute/off/switch/on/unmute-procedure without instant-switching capability either resets or randomizes start positions (relative to transients). The former case puts harder burden on the subject's capabilities, the latter adds random noise, which must be accommodated by increasing the number of rounds. So ABX testing without the hassle of time-synching doesn't seem fundamentally flawed, but only shifts effort from technical to procedural expenditure. When properly conducted the probability of false positives can be equal with both approaches. Or am I overlooking something?

(Not a) good explanation of jitter in TAS

Reply #123
To tell the whole story, if nearly instant switching is not available, then the potential sensitivity of the comparison is seriously, perhaps even debilitatingly compromised.


A mute/off/switch/on/unmute-procedure without instant-switching capability either resets or randomizes start positions (relative to transients). The former case puts harder burden on the subject's capabilities, the latter adds random noise, which must be accommodated by increasing the number of rounds.


In both cases the listener is put at a disadvantage. IOW, if nearly instant, transient-free switching is not available, then the potential sensitivity of the comparison is seriously, perhaps even debilitatingly compromised.

To me there is only one solution - get the switching as near-instant, and transient free as you can. 


Quote
So ABX testing without the hassle of time-synching doesn't seem fundamentally flawed,


That's a decision you get to make. I see you falling into the golden-ear's hands by feeding their prejudices that we bias our tests against  possible positive outcomes for the listener.





(Not a) good explanation of jitter in TAS

Reply #124
I see you falling into the golden-ear's hands by feeding their prejudices that we bias our tests against  possible positive outcomes for the listener.


That depends on the perspective. For significant positive results you don't necessarily need time-synched switching. Just accomodate the procedure: either reset positions or randomize (+/- several frames) and increase the number of trials. Both can lead to solid positive, but weaker negative results.

Time-synching increases the probability of true positive results (per round), thus strengthens the significance of negative results somewhat. But if you're just after possible positive results between DACs - what this thread was about recently - demanding time-synching* is not necessary and can only improve chances to find something, when proper non time-synched methods have failed to reveal a difference.


(* instead of simple mute/off/switch/on/unmute-switching)

 
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