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Topic: (Not a) good explanation of jitter in TAS (Read 88757 times) previous topic - next topic
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(Not a) good explanation of jitter in TAS

Reply #175
SRC is a very carefully designed issue in OS X. The only limitation is that you can't playback two sample rates into the same (virtual) device at once without SRC, which makes sense. Different rates for input and output or completely different devices is no problem (as long as your hardware supports this). You can even choose the clock source (when available) on a per device level through the standard OS dialogs.

I think the >22 kHz content you are seeing is HF noise, not aliasing. To be sure I am going to repeat the test later with everything at 96 kHz. Initially I had chosen the approach to playback at 44.1 kHz because the DACs' Redbook performance is what I was mainly interested in. Recording was chosen to be 24/96 to minimize the ADC's influence on the results.

(Not a) good explanation of jitter in TAS

Reply #176
ABX results

foo_abx 1.3.4 report
foobar2000 v0.9.6.8
2009/08/12 09:29:00

File A: \\bmsbs\redirectedfolders\JSiau\Desktop\ABX Tests\DAC1 vs MacBookPro\rec_dac1_24_corrected.flac
File B: \\bmsbs\redirectedfolders\JSiau\Desktop\ABX Tests\DAC1 vs MacBookPro\rec_mbp_24.flac

09:29:00 : Test started.
09:32:33 : 01/01  50.0%
09:33:04 : 02/02  25.0%
09:33:18 : 03/03  12.5%
09:33:27 : 04/04  6.3%
09:33:46 : 05/05  3.1%
09:33:57 : 06/06  1.6%
09:34:06 : 07/07  0.8%
09:34:22 : 08/08  0.4%
09:34:30 : 09/09  0.2%
09:34:41 : 10/10  0.1%
09:34:50 : 11/11  0.0%
09:34:57 : 12/12  0.0%
09:35:09 : 13/13  0.0%
09:35:19 : 14/14  0.0%
09:35:35 : 15/15  0.0%
09:35:44 : 16/16  0.0%
09:35:54 : Test finished.

----------
Total: 16/16 (0.0%)


The rec_mbp_24 file sounds like it was truncated to 16-bits somewhere in the processing chain.  This truncation causes noise pumping and distortion on the fade out of the final note.

Used a DAC1 USB for playback using the USB input and headphone output.  Headphones were closed-back Sony MDR-V600.
John Siau
Vice President
Benchmark Media Systems, Inc.

(Not a) good explanation of jitter in TAS

Reply #177
Would it be fair to say that you turned the volume up to listen to the fade out?

(Not a) good explanation of jitter in TAS

Reply #178
I had the volume up quite far, but I took care that the whole normalized track was still listenable without getting too uncomfortable and I did not turn up the volume higher for the 0:07-0:09 position.

Why there should be 16 bit truncation going on, is a riddle to me. * From the point of any user controllable parameters I can rule it out, but I can't guarantee the same for the hardware and driver level. I wish I had better equipment available right now. I'm going to prepare a second round of testing in a minute and will upload a new set of samples afterwards.

PS * Maybe the input level gain control is digital instead of analog. I'll try identical input gain levels in the second round.

(Not a) good explanation of jitter in TAS

Reply #179
Would it be fair to say that you turned the volume up to listen to the fade out?

Yes.  The first test just confirms that an artifact exists.  It does not address the audibility in normal listening.

I am just about to re-run the test at a lower volume using speakers.  I will do two tests:  The first test will be in my office where the AC noise is measuring 39 to 41 dBA slow.  The second test will be in our listening room where the ambient noise is below the range of my B&K 2219 Sound Level Meter. I will measure and document the SPL produced by a  1 kHz -20 dB FS test tone (at the same settings used in the tests).
John Siau
Vice President
Benchmark Media Systems, Inc.

(Not a) good explanation of jitter in TAS

Reply #180
foo_abx 1.3.4 report
foobar2000 v0.9.6.8
2009/08/12 11:58:23

File A: \\bmsbs\redirectedfolders\JSiau\Desktop\ABX Tests\DAC1 vs MacBookPro\rec_dac1_24_corrected.flac
File B: \\bmsbs\redirectedfolders\JSiau\Desktop\ABX Tests\DAC1 vs MacBookPro\rec_mbp_24.flac

11:58:23 : Test started.
11:58:47 : 01/01  50.0%
11:58:58 : 02/02  25.0%
11:59:05 : 03/03  12.5%
11:59:11 : 04/04  6.3%
11:59:19 : 05/05  3.1%
11:59:24 : 06/06  1.6%
11:59:30 : 07/07  0.8%
11:59:36 : 08/08  0.4%
11:59:42 : 09/09  0.2%
11:59:48 : 10/10  0.1%
11:59:53 : 11/11  0.0%
11:59:58 : 12/12  0.0%
12:00:03 : 13/13  0.0%
12:00:15 : 14/14  0.0%
12:00:20 : 15/15  0.0%
12:00:28 : 16/16  0.0%
12:00:34 : Test finished.

----------
Total: 16/16 (0.0%)


Ambient noise at listening position 37 dBA (at my office desk)
Listening level: -20 dBFS test tone plays at 76 dBA at the listening position - both speakers driven
Test file measures 76 to 79 dBA slow at loudest section
D/A = DAC1 USB
Speakers/amplifier = Dynaudio BM5A (powered speakers)

Notes:  I am hearing a difference in the noise floor.  The MBP sample has higher noise and noise modulation.
John Siau
Vice President
Benchmark Media Systems, Inc.

(Not a) good explanation of jitter in TAS

Reply #181
foo_abx 1.3.4 report
foobar2000 v0.9.6.8
2009/08/12 14:47:44

File A: \\bmsbs\redirectedfolders\JSiau\Desktop\ABX Tests\DAC1 vs MacBookPro\rec_dac1_24_corrected.flac
File B: \\bmsbs\redirectedfolders\JSiau\Desktop\ABX Tests\DAC1 vs MacBookPro\rec_mbp_24.flac

14:47:44 : Test started.
14:48:13 : 01/01  50.0%
14:48:21 : 02/02  25.0%
14:48:28 : 03/03  12.5%
14:48:35 : 04/04  6.3%
14:48:41 : 05/05  3.1%
14:48:48 : 06/06  1.6%
14:48:55 : 07/07  0.8%
14:49:07 : 08/08  0.4%
14:49:17 : 09/09  0.2%
14:49:26 : 10/10  0.1%
14:49:37 : 11/11  0.0%
14:49:43 : 12/12  0.0%
14:49:51 : 13/13  0.0%
14:49:59 : 14/14  0.0%
14:50:06 : 15/15  0.0%
14:50:12 : 16/16  0.0%
14:50:20 : Test finished.

----------
Total: 16/16 (0.0%)

Ambient noise at listening position less than 30 dBA (in our listening room)
Listening level: -20 dBFS test tone plays at 64 dBA at the listening position - both speakers driven
Test file measures 69 to 71 dBA slow at loudest section
D/A = DAC1 USB
Speakers/amplifier = Klien and Hummel O 300D (powered speakers) - using analog inputs

Notes:  I am hearing a difference in the noise floor.  The MBP sample has higher noise and noise modulation.
John Siau
Vice President
Benchmark Media Systems, Inc.

(Not a) good explanation of jitter in TAS

Reply #182
I recall all posted samples and claims! 

Turns out, the difference was the ADC and test setup, not the DACs!

The DAC1 has much higher output volume in its standard "calibrated" setting than the MBP. Because of that I had adjusted the input gain setting in OS X's audio control panel in the first round until I had about equal values for both inputs on the level meter.

My ABX results were honestly much better than I ever would have expected, so I wanted to make sure that I hadn't overlooked anything for the second round. So this time I set the input gain to "0db" for both. I put the DAC1 into "variable" mode so that I could use its analog gain control to adjust its volume to exactly the same level as the MBP's output. I also converted the tec_sqam sample to 96 kHz prior to to playback as suggested (Izotope intermediate phase) and applied 24 bit dither after normalizing.

The results are like night and day now and both very good - only ABXable vs. the original at insane volume levels to me.

Two findings for MBP owners:

  • Better leave the "input gain control" untouched. It added considerable amounts of noise in my case.
  • Even if OS X allows different settings for input and output sample rate in the audio/midi control panel for the MBP, you get aliasing if you actually use it.


Sorry for the stir.

(Not a) good explanation of jitter in TAS

Reply #183
I recall all posted samples and claims! 

Turns out, the difference was the ADC and test setup, not the DACs!
Two findings for MBP owners:
  • Better leave the "input gain control" untouched. It added considerable amounts of noise in my case.
  • Even if OS X allows different settings for input and output sample rate in the audio/midi control panel for the MBP, you get aliasing if you actually use it.

Thanks for the correction.

What you have demonstrated is that digital volume control can greatly reduce the performance of computer sound cards.  I discussed this at the end of a prior posting: "Regarding the issue of Luxury"

Higher D/A (and A/D) performance is needed when digital volume controls are used.  For example, if 20 dB of digital attenuation is applied at normal listening levels, the D/A would need to have a signal to noise ratio (SNR) of about 116 dB to deliver CD quality.  Obviously a simple solution is to reduce the digital attenuation and apply some analog gain reduction.  Unfortunatly many consumer audio systems have no gain control in the analog-domain.  As a result, these systems often deliver audio that cannot approach CD quality.

rpp3po took great to try to get the best possible quality for the ABX source files.  The results were not what was expected because of the presence of the digital volume control.  This demonstrates just how difficult it can be to get good audio out of many consumer devices.

The ABX tests in the prior postings demonstrated that the digital volume control in the Mac Book Pro degraded the performance of its A/D converter to the extent that it was easy to hear differences at normal listening levels.  This degradation was significant enough to allow me to successfully ABX the two test files using the speakers and sound card built into my HP Mini 2133 netbook while listening in relatively high ambient noise levels produced by a nearby AC ventilator (44 dBA at the listening position):

foo_abx 1.3.4 report
foobar2000 v0.9.6.8
2009/08/14 10:25:42

File A: \\bmsbs\redirectedfolders\JSiau\Desktop\ABX Tests\DAC1 vs MacBookPro\rec_dac1_24_corrected.flac
File B: \\bmsbs\redirectedfolders\JSiau\Desktop\ABX Tests\DAC1 vs MacBookPro\rec_mbp_24.flac

10:25:42 : Test started.
10:26:15 : 01/01  50.0%
10:26:21 : 02/02  25.0%
10:26:27 : 03/03  12.5%
10:26:33 : 04/04  6.3%
10:26:39 : 05/05  3.1%
10:26:51 : 06/06  1.6%
10:26:59 : 06/07  6.3%
10:27:13 : 07/08  3.5%
10:27:28 : 08/09  2.0%
10:28:06 : 09/10  1.1%
10:28:12 : 10/11  0.6%
10:28:33 : 11/12  0.3%
10:28:49 : 11/13  1.1%
10:29:25 : 12/14  0.6%
10:29:34 : 13/15  0.4%
10:29:42 : 14/16  0.2%
10:29:54 : Test finished.

----------
Total: 14/16 (0.2%)

Ambient noise 44 dBA slow
Peak levels in sound track 79 to 81 dBA slow
Listening level: -20 dBFS test tone plays at 76 dBA at the listening position - both speakers driven
Volume control on HP Mini set to maximum
Playback through HP Mini internal sound card and internal speakers.
John Siau
Vice President
Benchmark Media Systems, Inc.

(Not a) good explanation of jitter in TAS

Reply #184
The problem I generally see with externally generated jitter, either through up-/downsampling or electronically, is the applicability of those results. You can take it so far, that you can hear at least something, i. e. what jitter of model m at gain g does or does not sound like. After that you will have several pairs m, g that seem to be relevant. But those then have to be translated and tested against real world implementations.


The way I see to address that situation is to simply know what kinds of jitter exist in the real world, and duplicate them.  I learned quite a bit about jitter by testing equipment for my now-defunct www.pcavtech.com web site. The most common kind of really heavy jitter is related to the power line frequency.  Frame rates also show up fairly frequently. While quite a bit has been written about self-jitter, it is not all that common at levels that are likely to be heard.

(Not a) good explanation of jitter in TAS

Reply #185
The problem I generally see with externally generated jitter, either through up-/downsampling or electronically, is the applicability of those results. You can take it so far, that you can hear at least something, i. e. what jitter of model m at gain g does or does not sound like. After that you will have several pairs m, g that seem to be relevant. But those then have to be translated and tested against real world implementations.


The way I see to address that situation is to simply know what kinds of jitter exist in the real world, and duplicate them.  I learned quite a bit about jitter by testing equipment for my now-defunct www.pcavtech.com web site. The most common kind of really heavy jitter is related to the power line frequency.  Frame rates also show up fairly frequently. While quite a bit has been written about self-jitter, it is not all that common at levels that are likely to be heard.


That's part of the problem though: it's hard to know what jitter really does to the signal. I have read a few papers on modelling jitter in the last days, and yet am not exactly sure what is going on. I suppose because it presumably depends on the DAC, which is hard to model. Most papers I've looked at do give explicit models, but they appear to have assumptions on jitter that I find it hard to take on trust (not really, I am sure it is accurate, but with no personal experience of what it "sounds like", they just seem to me to be assertions; I'd like to find a way to avoid those).

I'm running some numerical experiments right now but am not at all convinced by my techniques (lots of perhaps unwarranted assumptions) so will shut up for a while.

(Not a) good explanation of jitter in TAS

Reply #186
The problem I generally see with externally generated jitter, either through up-/downsampling or electronically, is the applicability of those results. You can take it so far, that you can hear at least something, i. e. what jitter of model m at gain g does or does not sound like. After that you will have several pairs m, g that seem to be relevant. But those then have to be translated and tested against real world implementations.


The way I see to address that situation is to simply know what kinds of jitter exist in the real world, and duplicate them.  I learned quite a bit about jitter by testing equipment for my now-defunct www.pcavtech.com web site. The most common kind of really heavy jitter is related to the power line frequency.  Frame rates also show up fairly frequently. While quite a bit has been written about self-jitter, it is not all that common at levels that are likely to be heard.


That's part of the problem though: it's hard to know what jitter really does to the signal.


It is hard to know the details of just about anything. That's why education can take so long and require so much effort,  There's another post active here that wants to know the flow of data from the CD to the headphone jack, including all data busses, all buffers, all data formats. Not a bad task for a thesis project.

Quote
I have read a few papers on modelling jitter in the last days, and yet am not exactly sure what is going on.


That would be your problem!

Quote
I suppose because it presumably depends on the DAC, which is hard to model.


Modeling DACs is very easy compared to modelling say, a vacuum tube or magnetic tape.

I don't think that a person needs to model DACs to understand jitter. Jitter is FM distortion or FM modulation. What do you know about FM modulation, or modulation in general?

Quote
Most papers I've looked at do give explicit models, but they appear to have assumptions on jitter that I find it hard to take on trust (not really, I am sure it is accurate, but with no personal experience of what it "sounds like", they just seem to me to be assertions; I'd like to find a way to avoid those).


The way to avoid many kinds of assertions is to gather evidence. To paraphrase what I said above, I have gathered a certain amount of evidence. I gathered evidence about modulation and I gathered evidence about the results of modulation in real world audio equipment. I have gathered information about how the ear perceives the results of modulation on audio signals. So much for many spare moments and also dedicated moments during maybe 15 years of my  life.

You lookin' for a rose garden, kid? ;-)

Quote
I'm running some numerical experiments right now but am not at all convinced by my techniques (lots of perhaps unwarranted assumptions) so will shut up for a while.


If you have questions, then by all means don't shut up about your questions. Ask them. I had virtually nobody to talk to when I was gathering my evidence. Very few people would even tell me where or how to gather evidence. I spent a lot of time and money measuring things and then analyzing what I measured. I had to buy expensive equipment. I had to buy expensive books.  Some people lied to me, outright. Others belittled me in public for asking questions.  Other people shared what I now know, based on superior evidence, to be wrong ideas. I probably have a few things wrong, too.

Welcome to the real world!

At least Amazon sells books to everybody with money!

(Not a) good explanation of jitter in TAS

Reply #187
My goal is to design products where all of the artifacts produced by the product fall below the threshold of hearing in a silent room.  In order to guarantee inaudibility, I am consciously ignoring masking effects and designing as if the only sound being played was that of the artifacts.  To me this means keeping the sum total of all artifacts at a level that is at least 110 dB below the peak audio levels.  This goal has been achievable in the analog domain for at least 20 years, and for the last 10 years has been achievable in the digital domain.  Achieving these goals requires more engineering, careful circuit layout, and a few more components.  We deliver products that meet these goals in a price range that may seem extravagant to the average consumer ($1000 to $2000).  But, these are very reasonable prices for professional products that will be used in a recording studio on a regular basis.  The hi-end hi-fi enthusiast looks at our products and says "how can it be any good if it only costs $2000"?


I don't see any compelling logic here.

If your goal was to design products where all of the artifacts produced by the artifact fall below the threshold of hearing in a silent room, any reasonable person would say that you are tilting at windmills, given that there are no silent rooms.

A reasonable goal is to design products where no real world customer will hear artifacts in the most silent room that he is likely ever to encounter. A resasonable goal is to design products where no real world customer will hear artifacts while wearing the most isolating earphones in the best room that he is likely ever to encounter.

Once you've adopted a reasonable goal, then you should use a reasonable means to define the details of that goal.

Saying that your goal is "A", and then immediately and without any other evidence spouting some arbitrary technical specifications is again, not at all reasonable.

Once you have stated your goal, a reasonable way to proceed would be to model the performance of a product that is just barely noticable in terms of all known audible artifacts.  Once you have produced that model, pick a reasonble but somewhat arbitrary number, and set your goal to produce a product that measures that good. For example, if you know all about the audibility of noise and distoriton, then set a goal that is X dB better than that.

Quote
The average consumer has no idea how bad their CD player or sound card really is.  There often are no published specifications.


This statement shows obvious prejudice and bias. An unbiased person would say:

The average consumer has no idea how good or bad their CD player or sound card really is.  They don't know what kind of specifications are required for total freedom from audible artifacts under critical but real  world listening conditions, and most product spec sheets are so incomplete that they don't know how the product compares to the actual requirements.



(Not a) good explanation of jitter in TAS

Reply #188
The average consumer has no idea how good or bad their CD player or sound card really is.  They don't know what kind of specifications are required for total freedom from audible artifacts under critical but real  world listening conditions, and most product spec sheets are so incomplete that they don't know how the product compares to the actual requirements.


I absolutely agree with you - well said!
John Siau
Vice President
Benchmark Media Systems, Inc.

(Not a) good explanation of jitter in TAS

Reply #189
My goal is to design products where all of the artifacts produced by the product fall below the threshold of hearing in a silent room.  In order to guarantee inaudibility, I am consciously ignoring masking effects and designing as if the only sound being played was that of the artifacts.  To me this means keeping the sum total of all artifacts at a level that is at least 110 dB below the peak audio levels.  This goal has been achievable in the analog domain for at least 20 years, and for the last 10 years has been achievable in the digital domain.  Achieving these goals requires more engineering, careful circuit layout, and a few more components.  We deliver products that meet these goals in a price range that may seem extravagant to the average consumer ($1000 to $2000).  But, these are very reasonable prices for professional products that will be used in a recording studio on a regular basis.  The hi-end hi-fi enthusiast looks at our products and says "how can it be any good if it only costs $2000"?


I don't see any compelling logic here.

If your goal was to design products where all of the artifacts produced by the artifact fall below the threshold of hearing in a silent room, any reasonable person would say that you are tilting at windmills, given that there are no silent rooms.

A reasonable goal is to design products where no real world customer will hear artifacts in the most silent room that he is likely ever to encounter. A resasonable goal is to design products where no real world customer will hear artifacts while wearing the most isolating earphones in the best room that he is likely ever to encounter.


Recording studios have many audio devices in cascade.  The system performance is limited by the combined performance of all of the devices.  My goal is to provide tools that don't get in the way (by causing audible defects).  A comfortable margin of safety is appreciated in a professional environment.  The goal of the recordist is to capture a musical performance.  If he is distracted by the poor performance of his equipment chain, he may miss the opportunity to capture that very special musical performance.

Obviously most consumers do not need or demand this sort of performance.
John Siau
Vice President
Benchmark Media Systems, Inc.

(Not a) good explanation of jitter in TAS

Reply #190
My goal is to design products where all of the artifacts produced by the product fall below the threshold of hearing in a silent room.  In order to guarantee inaudibility, I am consciously ignoring masking effects and designing as if the only sound being played was that of the artifacts.  To me this means keeping the sum total of all artifacts at a level that is at least 110 dB below the peak audio levels.  This goal has been achievable in the analog domain for at least 20 years, and for the last 10 years has been achievable in the digital domain.  Achieving these goals requires more engineering, careful circuit layout, and a few more components.  We deliver products that meet these goals in a price range that may seem extravagant to the average consumer ($1000 to $2000).  But, these are very reasonable prices for professional products that will be used in a recording studio on a regular basis.  The hi-end hi-fi enthusiast looks at our products and says "how can it be any good if it only costs $2000"?


I don't see any compelling logic here.

If your goal was to design products where all of the artifacts produced by the artifact fall below the threshold of hearing in a silent room, any reasonable person would say that you are tilting at windmills, given that there are no silent rooms.

A reasonable goal is to design products where no real world customer will hear artifacts in the most silent room that he is likely ever to encounter. A resasonable goal is to design products where no real world customer will hear artifacts while wearing the most isolating earphones in the best room that he is likely ever to encounter.


Recording studios have many audio devices in cascade.


I addressed this issue in a portion of my post that you didn't quote for some reason. Didin't read it? I didn't make it clear enough?

Quote
Once you have stated your goal, a reasonable way to proceed would be to model the performance of a product that is just barely noticable in terms of all known audible artifacts. Once you have produced that model, pick a reasonble but somewhat arbitrary number, and set your goal to produce a product that measures that good. For example, if you know all about the audibility of noise and distoriton, then set a goal that is X dB better than that.


I have no problem with intelligent overkill. I get the requirements of the production environment because I do production all the time.

My other problem is with the way you seem to be doing overkill, and I addressed that too, and you didn't quote or respond to that either:

Quote
Saying that your goal is "A", and then immediately and without any other evidence spouting some arbitrary technical specifications is again, not at all reasonable.


The point is that spouting some arbitrary technical specs isn't a logical way to proceed. The logical way to proceed is to determine the actual thresholds by means of reliable subjective tests,  come up with a reasonable overkill factor for each of them, and built to that balanced overkill design.

I've done a fair amount of regression testing to determine how various forms of distortion build up when euqipment is cascaded. Have you?






(Not a) good explanation of jitter in TAS

Reply #191
The point is that spouting some arbitrary technical specs isn't a logical way to proceed. The logical way to proceed is to determine the actual thresholds by means of reliable subjective tests,  come up with a reasonable overkill factor for each of them, and built to that balanced overkill design.
Since this thread is about jitter, let's have a look at a Benchmark statement about jitter:
Quote
Jitter can only be considered totally inaudible if the worst case jitter induced sidebands are at least 23 dB below the A-weighted system noise. Above this level jitter may be audible or it may be masked by the program audio. At Benchmark our goal is to achieve totally inaudible levels of jitter.
[/size]Any comments on that design goal ?

(Not a) good explanation of jitter in TAS

Reply #192
Quote
Jitter can only be considered totally inaudible if the worst case jitter induced sidebands are at least 23 dB below the A-weighted system noise. Above this level jitter may be audible or it may be masked by the program audio. At Benchmark our goal is to achieve totally inaudible levels of jitter.
[/size]Any comments on that design goal ?
Not compliant with TOS8. Here, jitter can only be considered totally inaudible if someone fails an ABX test searching for it. But you can't prove a negative, so the closest we can come is finding the threshold at which jitter becomes audible and assuming that it is inaudible below that threshold.

(Not a) good explanation of jitter in TAS

Reply #193
The point is that spouting some arbitrary technical specs isn't a logical way to proceed. The logical way to proceed is to determine the actual thresholds by means of reliable subjective tests,  come up with a reasonable overkill factor for each of them, and built to that balanced overkill design.
Since this thread is about jitter, let's have a look at a Benchmark statement about jitter:
Quote
Jitter can only be considered totally inaudible if the worst case jitter induced sidebands are at least 23 dB below the A-weighted system noise. Above this level jitter may be audible or it may be masked by the program audio. At Benchmark our goal is to achieve totally inaudible levels of jitter.
[/size]Any comments on that design goal ?


Besides the obvious TOS8 violation, it fails reasonableness checks. By setting the acceptable level for jitter based simply on a level, it ignores the fact that masking is strongly dependent on the spectral distance between the spurious response(s) due to jitter and the masking signal(s). It also ignores the fact that the human ear is far more sensitive to FM distortion at some frequencies than others.

To put the audiblity issues related to jitter, we must remember that both the LP format and analog tape are simply full of jitter, and that the entire high end industry has nothing to say about it. I don't think there ever has been a halfways decent CD player that had even 1/100th the jitter inherent in LP playback or analog tape. Furthermore some of the early CD players that high end audiophiles love to hate such as the CDP 101, actually have very low jitter.


Obviously, jitter is one of high end audio's more obvious red herrings.

(Not a) good explanation of jitter in TAS

Reply #194
the closest we can come is finding the threshold at which jitter becomes audible and assuming that it is inaudible below that threshold.
From reading the Benchmark paper I had the impression that it was exactly that what they did. They didn't specify their sources (scientific papers, their own DBT's?) but the assumption is that when tones (in this case clock jitter distortion) are at least 23 dB below the system (in this case ADC) noise, they can be assumed inaudible.
Do you have any evidence that this assumption is incorrect ?
Obviously, jitter is one of high end audio's more obvious red herrings.
Your TOS8 violation indicates that you have doubts about the Benchmark claim of inaudibility of the jitter effects. On the other hand you state that any halfway CD player has much better jitter performance than LP or analogue tape, which is concidered (proven?) to be no audible problem.
In that case it's probably not unreasonable to assume that the Benchmark jitter performance is far below audibility thresholds ?

(Not a) good explanation of jitter in TAS

Reply #195
Obviously, jitter is one of high end audio's more obvious red herrings.
Your TOS8 violation indicates that you have doubts about the Benchmark claim of inaudibility of the jitter effects. On the other hand you state that any halfway CD player has much better jitter performance than LP or analogue tape, which is concidered (proven?) to be no audible problem.
In that case it's probably not unreasonable to assume that the Benchmark jitter performance is far below audibility thresholds ?


I'm pretty sure that the jitter in just about anything halfways decent is inaudible.

There's a JAES paper that covers this:

"Theoretical and Audible Effects of Jitter on Digital Audio Quality", Benjamin, Eric and Gannon, Benjamin, 105th AES Convention, 1998, Preprint 4826.

I don't know if the samples mentioned here are still acessible:

HDDaudio jitter test files link