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post #1 of 284 Old 05-11-2014, 09:17 PM - Thread Starter
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Continuing discussion from the closed "optical cable or hdmi" thread.

amir,
I'm still having a hard time appreciating that the level of hdmi jitter measured (-90dB) in the Anthem AV Receiver output is audible in a real world listening environment. Particularly when we consider the proximity of distortion to the fundamental tone as JHAZ deftly noted (within one hertz). As we agreed, only at reference SPL's does the associated distortion reach a whisper level.

Fielder's study that system noises could be heard as much as 15dB below that of room noise was noted. Are there any studies that demonstrate a 15db tone at 11,875Hz can be heard when a 105dB 12,000Hz tone is present? Or for the same frequencies, a 0dB noise can be heard with 90db room noise?

If we return to the NwAvGuy website, he cites -100dB as the audible threshold for jitter, and, judging by the information presented and citations, he doesn't appear to have low standards or a small grasp of the subject.

http://nwavguy.blogspot.com/2011/02/jitter-does-it-matter.html
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post #2 of 284 Old 05-11-2014, 09:51 PM - Thread Starter
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Its hard to digest that players are rendering audible levels of distortion given the following sources.

Audioholics Oppo player measurements and comments:

http://www.audioholics.com/blu-ray-and-dvd-player-reviews/copy_of_oppo-bdp-93/oppo-bdp-93-bdp-95-on-the-bench

"We ran both 192kHz/24 bit 6 channel and 96kHz/ 24 bit 8 channel Dolby TrueHD signals into both players and both produced similarly ruler flat frequency performance from 20Hz all the way up to the Audio Precision test gear bandwidth limitation (80kHz).

We also ran Bit Error Rate (BER) tests on both players using Dolby TrueHD and DTS HD test signals and both players produced a 0% BER which means both players delivered audio via HDMI 100% error free. Early HDMI products (ver 1.0 to 1.2) reportedly suffered from jitter related issues but HDMI ver 1.3 and above has completely eliminated jitter related issues thus ensuring bit for bit exact signal transfer from the player to the A/V receiver or processor."

Since the Oppo is highly regarded, I looked up specs for an inexpensive Sansa Clip $30 which measured less than 115dB jitter.

http://nwavguy.blogspot.com/2011/02/sansa-clip-measured.html


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post #3 of 284 Old 05-11-2014, 10:04 PM
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Thanks for creating this thread seeing how the other was closed. I am in the process of setting up my CNC machine which is a lot more fun than posting on forums smile.gif. I will come back when I get to a breaking point. Appreciate your patience and of course others can comment in my absence.

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post #4 of 284 Old 05-12-2014, 06:38 AM
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From my own experience I don't think HDMI jitter affects compressed audio streams (DTS, DD) somehow. Uncompressed PCM is another story.

Burned by the Audio Inquisition
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post #5 of 284 Old 05-12-2014, 11:07 AM
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Originally Posted by amirm View Post

......I am in the process of setting up my CNC machine .....

What do you have? I'm looking into this too.

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post #6 of 284 Old 05-12-2014, 01:51 PM
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Originally Posted by Glimmie View Post

What do you have? I'm looking into this too.
I bought a used machine. Wanted to be able to machine full size sheets (4x8) and new they cost upward of $20K. Found one on Craigslist from a shop that was upgrading to a larger machine at much lower cost. It has upgraded components which is good. Has closed loop servo, Hiwin slides, etc. The original company that made it went out of business. Fortunately there is not much to these things. Just the table, gantry and motors.

Looked a lot of other units but they all got quite pricey as I mentioned when the size went up. I bought this larger machine for the price of a new small one. I looked at CNC Shark (lots of complaints about this), Shop Bot (excellent support), and Camaster (pretty sturdy welded steel framing).

One of the big pain points was shipping. Even the lightest unit clocks around 1,000 pounds. Companies like Shopbot wanted $1,000 or so for shipping plus $300 for crating alone! Then the thing would arrive on a truck and was your problem getting it off and into your shop. The Shopbot came in pieces and takes two days to put it together.

Plan is to use it to make speaker enclosures, machined MDF (both for work) and furniture (for me personally). It is my first CNC so going the used route seemed like a very good bet assuming you do some legwork on what components are good. Suggest looking on Craigslist for a few days and see if anyone puts one up for sale.

Sorry for the off-topic posts guys smile.gif.

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post #7 of 284 Old 05-12-2014, 01:52 PM
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Quote:
Originally Posted by Qaq View Post

From my own experience I don't think HDMI jitter affects compressed audio streams (DTS, DD) somehow. Uncompressed PCM is another story.
It actually affects both equally, as non-intuitive as that might be.

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post #9 of 284 Old 05-13-2014, 04:29 AM
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amirm, you know what I mean. Compressed stream needs to be buffered to be decoding and that eluminates digital interface jitter.
A few years ago I've made two lossless music samples from one source - one packed to DTS-HD MA and another one packed to Flac. Just to check that my theory. 1st one was played as bitstream (compressed) and 2nd one as PCM (uncompressed). From the listening test (non blind) I prefered bitstreamed one. I still keep those samples BTW.

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post #10 of 284 Old 05-14-2014, 09:08 PM - Thread Starter
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Well hey, I can't compete with a new power tool. If I'm up against a battery operated screw driver or a hand held shop vac, and the discussion was about double-blind listening tests, then maybe my post would get more attention.

I've got this embarrassing problem that I probably shouldn't bring up in public, but since I began listening to my new disc player at high volumes, I begin to dribble. I know its foolish, because its not like I'm not looking at a really attractive woman or a juicy, flame-broiled steak.

So I went to my doctor and he asked me a lot of questions about my listening habits, and my audio equipment, and he says its jitter. Jitter induced dribble (JID). Poor DAC implementation, clocking issues, and all that electrical stuff. I asked if he was sure, could it be exposure to too much clipping, improper subwoofer phasing, or leaky amp transistors? He said "no," those things would result in profuse sweating, nausea, and sticky underwear.

So he told me to get rid of my inexpensive Monoprice hdmi cables right away, and gave me a prescription for some Monster cables, which ARE NOT covered by my HMO insurance plan, and said, its okay to use spdif for a while. Being the Yale man that he is, I thought he was full of it, until I read the comments about low jitter with spdif.

So I've got a follow up appointment at the end of the month, and if I'm still JID'ing, he's going to have to remove my new disc player, and put in something with a toroidal transformer, linear phase filters with 'constant group delay', and maybe even tube transistors and a 384kHz upsampling feature.

Just great! Thanks a lot AVS Forum!
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post #11 of 284 Old 05-15-2014, 07:09 AM
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That Dr. sounds like a quack. The solution to your ailment is very simple:


Floyd

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post #12 of 284 Old 05-15-2014, 07:58 AM
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Quote:
Originally Posted by amirm View Post

I bought a used machine. Wanted to be able to machine full size sheets (4x8) and new they cost upward of $20K. Found one on Craigslist from a shop that was upgrading to a larger machine at much lower cost. It has upgraded components which is good. Has closed loop servo, Hiwin slides, etc. The original company that made it went out of business. Fortunately there is not much to these things. Just the table, gantry and motors.

Looked a lot of other units but they all got quite pricey as I mentioned when the size went up. I bought this larger machine for the price of a new small one. I looked at CNC Shark (lots of complaints about this), Shop Bot (excellent support), and Camaster (pretty sturdy welded steel framing).

One of the big pain points was shipping. Even the lightest unit clocks around 1,000 pounds. Companies like Shopbot wanted $1,000 or so for shipping plus $300 for crating alone! Then the thing would arrive on a truck and was your problem getting it off and into your shop. The Shopbot came in pieces and takes two days to put it together.

Plan is to use it to make speaker enclosures, machined MDF (both for work) and furniture (for me personally). It is my first CNC so going the used route seemed like a very good bet assuming you do some legwork on what components are good. Suggest looking on Craigslist for a few days and see if anyone puts one up for sale.

Sorry for the off-topic posts guys smile.gif.

Servo?

Oh dear. still being on topic while off topic. tongue.gif

http://ieeexplore.ieee.org/xpl/login.jsp?tp=&arnumber=4504638&url=http%3A%2F%2Fieeexplore.ieee.org%2Fiel5%2F28%2F4504612%2F04504638.pdf%3Farnumber%3D4504638
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post #13 of 284 Old 05-15-2014, 08:31 AM
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You guys are really funny smile.gif.

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post #14 of 284 Old 05-15-2014, 08:58 AM
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Still busy with the CNC machine. But here is the first installment on the topic.

There is an unsaid aspect of our AV systems that plays a core role in this topic. That is, our Audio systems uses what is called a "push" method. That means that the source be it a DVD/BD player, Computer, etc. is the "master." It decides what to play naturally but also decides when the individual audio samples need to play. Ordinarily one assumes that the sampling rate, e.g. 44.1 Khz, would decide that. But that is not true. Simple reason is that the source data rate may drift from this speed to be slower or faster over time. And content can be encoded to not be at this exact rate (often the case in audio for video content).

The above means that in no case can the receiver/DAC assume it knows how fast the data is coming from the source and use its own “clock” to meter out the audio samples to the DAC chip. What it must do instead is track the input. If it slows down, it must slow down its own clock. If it speeds up, it must speed up accordingly. Not doing so means that over time it will lose sync with the source. In the case of audio for video, you would lose sync. And in all cases, you either run out of audio samples if the source is running slower, or wind up with too much data if the source is faster.

Another complication is the multiple sampling rates. Today’s AV/Audio products go from say 32 Khz all the way up to 192 Khz if not higher. The receiver then needs to detect this large range of “speeds” and track them in order to play at the same rate as the input.

Complicating matters is that while we think of these events, i.e. “clock” to be digital, this is not the case as soon as you put that signal on a wire, e.g. HDMI or S/PDIF. A square wave where we go from 0 to 1 instantly would require infinite bandwidth and no cable can provide that. As a result, what the receiver sees, is a distorted “digital” waveform. The distortion includes edges that are sloped instead of straight up, noisy, and waveforms that don’t resemble clean square pulses.

The solution to all of these problems is a circuit called Phased Locked Loop or PLL for short. In layman terms, we start with a clock rate of our own. We then compare that to the incoming clock. If our rate is lower, we keep increasing our rate until it matches the input. And the reverse if we are too fast. This necessitates a feedback loop much like you have in an amplifier.

Feedback circuits are simple on paper as I just described above. But in practice they can be quite tricky to get right. The most important issue is that once you connect the output to the input, you can get feedback much like a mic whistling that in a PA system. The solution is some amount of filtering. In this case, it means ignoring fast changes in the input rate. Here is the challenge: if you filter too much, then it takes you a long time to catch up to the input rate. In practical terms, if you switch inputs on your AVR or DAC, it may take it a while to start playing. If you don’t filter enough then some of those unwanted variations get through and that is what we call “jitter.”

Note that “buffering” p lays no role here. A buffer is a chunk of memory that temporarily holds audio samples. The standard assumption is that we can dump the input data into a buffer, then set our DAC clock to an accurate value, e.g. 44.1 KHz, and do away with jitter. By now you should know why that does not work. Repeating earlier explanation, if you did that, you would drift away from the source which is not allowed.

Buffering will be used to hold audio samples while the PLL tracks the input but it does nothing to eliminate jitter by itself.

Commenting on the notion of sending a bitstream to the receiver instead of PCM audio samples, the idea is to send the compressed stream to the AVR and have it decode it. The understanding is that if the AVR is decoding the bits, and what comes over HDMI is “data” you no longer have to worry about jitter induced by the source and HDMI cable/receiver. Unfortunately it doesn’t work that way. Yes, the DSP in the AVR will do the decoding. But no, it doesn’t get to make up its own fixed clock. The requirement still is to track the input. What this means that the AVR decodes the compressed stream but as it attempts to play them, it must track the input rate – not its own clock.

Can there be audible differences between decoding in the AVR vs in the source? Maybe. The activity in the AVR changes and therefore the profile of noise that gets induced into the DAC changes. So in theory there can be differences. But without measurements we don’t know. The expectation bias, should you assume that jitter goes away with bitstream mode is very high and hence subjective evaluations may not be valid.

Speaking of activity, that is what pollutes our DAC clock or its other sensitive circuits. It might seem that the DAC circuit is independent of the rest of the system so nothing should mess with it. Unfortunately that is not how the electronics work. For one thing, there is ultimately one power source for the whole system and noise and interference can travel through that path from one subsystem, e.g. noisy digital subsystem, to the “quiet” part which is the DAC. Worse yet, even without any connection interference can “couple” into the DAC. You no doubt know that we can transmit signals over the air. That is how your radio works. The same occurs inside your audio/video gear. In the case of HDMI there are a ton of circuits active due to sophistication of this interface. If the DAC is not fully protected against such sources of interference, its performance will vary based on whether you use HDMI or not.

Note that a self-enclosed system has a huge advantage over one that has its source and output DAC separate by wires. This explains why a portable audio player may have better performance than a Blu-ray player feeding an AVR. The former has no requirement at all to track any input. Its DAC can set its clock rate and then read music samples from its storage media as it needs it. This is the “pull” method and in almost all architectures is superior to our standard “push” method. In the Blu-ray player case we cannot set the clock and hence are at the mercy of what an interface like HDMI does to us.

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post #15 of 284 Old 05-15-2014, 02:02 PM
 
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Quote:
Originally Posted by amirm View Post

Still busy with the CNC machine. But here is the first installment on the topic.

There is an unsaid aspect of our AV systems that plays a core role in this topic. That is, our Audio systems uses what is called a "push" method. That means that the source be it a DVD/BD player, Computer, etc. is the "master." It decides what to play naturally but also decides when the individual audio samples need to play. Ordinarily one assumes that the sampling rate, e.g. 44.1 Khz, would decide that. But that is not true. Simple reason is that the source data rate may drift from this speed to be slower or faster over time. And content can be encoded to not be at this exact rate (often the case in audio for video content).

The above means that in no case can the receiver/DAC assume it knows how fast the data is coming from the source and use its own “clock” to meter out the audio samples to the DAC chip. What it must do instead is track the input. If it slows down, it must slow down its own clock. If it speeds up, it must speed up accordingly. Not doing so means that over time it will lose sync with the source. In the case of audio for video, you would lose sync. And in all cases, you either run out of audio samples if the source is running slower, or wind up with too much data if the source is faster.

Another complication is the multiple sampling rates. Today’s AV/Audio products go from say 32 Khz all the way up to 192 Khz if not higher. The receiver then needs to detect this large range of “speeds” and track them in order to play at the same rate as the input.

Complicating matters is that while we think of these events, i.e. “clock” to be digital, this is not the case as soon as you put that signal on a wire, e.g. HDMI or S/PDIF. A square wave where we go from 0 to 1 instantly would require infinite bandwidth and no cable can provide that. As a result, what the receiver sees, is a distorted “digital” waveform. The distortion includes edges that are sloped instead of straight up, noisy, and waveforms that don’t resemble clean square pulses.

The solution to all of these problems is a circuit called Phased Locked Loop or PLL for short. In layman terms, we start with a clock rate of our own. We then compare that to the incoming clock. If our rate is lower, we keep increasing our rate until it matches the input. And the reverse if we are too fast. This necessitates a feedback loop much like you have in an amplifier.

Feedback circuits are simple on paper as I just described above. But in practice they can be quite tricky to get right. The most important issue is that once you connect the output to the input, you can get feedback much like a mic whistling that in a PA system. The solution is some amount of filtering. In this case, it means ignoring fast changes in the input rate. Here is the challenge: if you filter too much, then it takes you a long time to catch up to the input rate. In practical terms, if you switch inputs on your AVR or DAC, it may take it a while to start playing. If you don’t filter enough then some of those unwanted variations get through and that is what we call “jitter.”

Note that “buffering” p lays no role here. A buffer is a chunk of memory that temporarily holds audio samples. The standard assumption is that we can dump the input data into a buffer, then set our DAC clock to an accurate value, e.g. 44.1 KHz, and do away with jitter. By now you should know why that does not work. Repeating earlier explanation, if you did that, you would drift away from the source which is not allowed.

Buffering will be used to hold audio samples while the PLL tracks the input but it does nothing to eliminate jitter by itself.

Commenting on the notion of sending a bitstream to the receiver instead of PCM audio samples, the idea is to send the compressed stream to the AVR and have it decode it. The understanding is that if the AVR is decoding the bits, and what comes over HDMI is “data” you no longer have to worry about jitter induced by the source and HDMI cable/receiver. Unfortunately it doesn’t work that way. Yes, the DSP in the AVR will do the decoding. But no, it doesn’t get to make up its own fixed clock. The requirement still is to track the input. What this means that the AVR decodes the compressed stream but as it attempts to play them, it must track the input rate – not its own clock.

Can there be audible differences between decoding in the AVR vs in the source? Maybe. The activity in the AVR changes and therefore the profile of noise that gets induced into the DAC changes. So in theory there can be differences. But without measurements we don’t know. The expectation bias, should you assume that jitter goes away with bitstream mode is very high and hence subjective evaluations may not be valid.

Speaking of activity, that is what pollutes our DAC clock or its other sensitive circuits. It might seem that the DAC circuit is independent of the rest of the system so nothing should mess with it. Unfortunately that is not how the electronics work. For one thing, there is ultimately one power source for the whole system and noise and interference can travel through that path from one subsystem, e.g. noisy digital subsystem, to the “quiet” part which is the DAC. Worse yet, even without any connection interference can “couple” into the DAC. You no doubt know that we can transmit signals over the air. That is how your radio works. The same occurs inside your audio/video gear. In the case of HDMI there are a ton of circuits active due to sophistication of this interface. If the DAC is not fully protected against such sources of interference, its performance will vary based on whether you use HDMI or not.

Note that a self-enclosed system has a huge advantage over one that has its source and output DAC separate by wires. This explains why a portable audio player may have better performance than a Blu-ray player feeding an AVR. The former has no requirement at all to track any input. Its DAC can set its clock rate and then read music samples from its storage media as it needs it. This is the “pull” method and in almost all architectures is superior to our standard “push” method. In the Blu-ray player case we cannot set the clock and hence are at the mercy of what an interface like HDMI does to us.


Nicely written - thank you for your quality time and effort in here.

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post #16 of 284 Old 05-16-2014, 08:14 PM - Thread Starter
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Many thanks Amir for your insightful comments. The challenges for the digital designer are apparent. Despite my position as a dac neophyte, from your well written explanations I'm deducing that producing accurate clocking in the player is decidedly easier than that of the receiver. The outstanding Oppo ($499/$1,200) and Sansa (>$40) player Jtest specs provide excellent examples of inaudible jitter rendered by the player and little need for jitter rejection by the receiver.

Thus far we have credible opinions indicating that -90db jitter in the AVR is of little consequence. I'll add another opinion from the following website:

http://archimago.blogspot.com/2013/10/measurements-onkyo-tx-nr1009-as-hdmi.html



The graphs above illustrate hdmi jitter under -100db. DUT is an onkyo-tx-nr1009 AV Receiver. The following observations are given by the Reviewer:

"Let's think about the J-Test and what was found in measuring the Onkyo for a moment. The Dunn J-Test is a synthetic test of data jitter first published by the late Julian Dunn around 1994 which (in the 24-bit 48kHz version) superimposes an undithered LSB 250Hz square wave over a primary 12kHz -3.01dBFS sine wave which is of course an exact 1/4 of the sampling rate. This superimposition stimulates the effect of subtle timing inaccuracies (jitter) which can be demonstrated as accentuation of the sidebands measured in the spectral graphs.

Remember that this test is synthetic and stimulative. What you see measured is not something you're probably ever going to "hear" in real music! The noise floor is not going to be down to the last bit in 16-bit audio and essentially impossible with recorded 24-bit audio (unless it's purely computer synthesized music). Also, jitter is more pronounced in the higher frequencies (11kHz and 12kHz are used as the primary signals in the J-Test). Realize that the human hearing sensitivity is well on its way down by 5kHz (as can be seen by the Fletcher-Munson Curves). Furthermore, if we specifically look at the Onkyo's J-test spectrum, the most pronounced side bands are about -90dB below the primary signal. To make matter even less worrisome is that the tall sidebands are all +/-250Hz around the primary signal and the audibility would be masked even if one did have awesome auditory acuity at 11/12kHz and could hear a signal 90dB down! This is also why I feel adding up all those sideband peaks and calling it a number (whatever picosecond or nanosecond) is really not all that useful when it comes to audibility."
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post #17 of 284 Old 05-16-2014, 08:17 PM - Thread Starter
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That Dr. sounds like a quack. The solution to your ailment is very simple:


fotto,

The garment is almost 'to-die-for,' wouldn't you say? Its a little plain, which is nothing a bag of sequins and a glue gun can't take care of. It appears to be very well made, thick, and much better designed for soaking up drool than a shirt sleeve. The pocket on the lower edge is very... very... nice. Why, a guy could empty a bag of snacks in there, popcorn or peanuts, and still have room for a small mending kit. You know, a needle and thread, scissors, and nail clippers. When you're done snipping just brush the clippings into the pocket. Ingenious.
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post #18 of 284 Old 05-17-2014, 05:40 AM
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Indeed! A MUST HAVE for those extended periods of critical listening. Why, I even let me wee dog ride in the pouch every morning while walking out to get the paper, and it becomes self cleaning to boot.

Floyd

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post #19 of 284 Old 05-17-2014, 08:29 AM
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Thus far we have credible opinions indicating that -90db jitter in the AVR is of little consequence.
Thank you for getting to the bottom of it. It's the important point which Mr. Salesman tries to hide. Why? For sales of course. wink.gif
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post #20 of 284 Old 05-17-2014, 11:07 AM
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Originally Posted by UndersAVS View Post

Many thanks Amir for your insightful comments.
My pleasure. Good to hear that my ramblings make sense to some people smile.gif.
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The challenges for the digital designer are apparent. Despite my position as a dac neophyte, from your well written explanations I'm deducing that producing accurate clocking in the player is decidedly easier than that of the receiver.
Exactly. In an odd turn of event, a "home theater in a box" would be a superior architecture to our separate boxes. Unfortunately the target for that market is lowest cost and so the optimization that would exist is not exploited.
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Thus far we have credible opinions indicating that -90db jitter in the AVR is of little consequence.
I am sorry I must have missed the credible opinions you are referring to. Which one is that?
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I'll add another opinion from the following website:

http://archimago.blogspot.com/2013/10/measurements-onkyo-tx-nr1009-as-hdmi.html



The graphs above illustrate hdmi jitter under -100db. DUT is an onkyo-tx-nr1009 AV Receiver. The following observations are given by the Reviewer:

"Let's think about the J-Test and what was found in measuring the Onkyo for a moment. The Dunn J-Test is a synthetic test of data jitter first published by the late Julian Dunn around 1994 which (in the 24-bit 48kHz version) superimposes an undithered LSB 250Hz square wave over a primary 12kHz -3.01dBFS sine wave which is of course an exact 1/4 of the sampling rate. This superimposition stimulates the effect of subtle timing inaccuracies (jitter) which can be demonstrated as accentuation of the sidebands measured in the spectral graphs.
While his explanation of what the J-test is, his reasoning for why it is used at the end is incorrect. The file itself contains no jitter whatsoever. Nor does it simulate "subtle timing inaccuracies." Remember, a file sitting on your hard disk before being played is just a bunch of digital audio samples. it is "perfect" in that regard. We can copy it 100 times and the last copy will be identical to first. So no timing problem exists or can exist in a digital file on a computer.

The J-test signal is designed to cause more jitter on the S/PDIF link. Due to the way the clock is transmitted on that link, the J-test signal can excite them due to the way the digital values are picked in that test signals. Interestingly enough, it does not do that for HDMI. Fact that HDMI does much worse without being "aggravated" points to serious issues underlying its implementation.

Don has written an excellent deep dive into jitter that gets induced on digital cables and is worth a read.
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Remember that this test is synthetic and stimulative. What you see measured is not something you're probably ever going to "hear" in real music! The noise floor is not going to be down to the last bit in 16-bit audio and essentially impossible with recorded 24-bit audio (unless it's purely computer synthesized music). Also, jitter is more pronounced in the higher frequencies (11kHz and 12kHz are used as the primary signals in the J-Test). Realize that the human hearing sensitivity is well on its way down by 5kHz (as can be seen by the Fletcher-Munson Curves). Furthermore, if we specifically look at the Onkyo's J-test spectrum, the most pronounced side bands are about -90dB below the primary signal. To make matter even less worrisome is that the tall sidebands are all +/-250Hz around the primary signal and the audibility would be masked even if one did have awesome auditory acuity at 11/12kHz and could hear a signal 90dB down! This is also why I feel adding up all those sideband peaks and calling it a number (whatever picosecond or nanosecond) is really not all that useful when it comes to audibility."
There is merit in this which once I get more time smile.gif, I will explain.

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post #21 of 284 Old 05-18-2014, 11:03 AM - Thread Starter
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Hello amir,
I find your ramblings consistently sensible wink.gif An exception when you wrote this sentence: "While his explanation of what the J-test is, his reasoning for why it is used at the end is incorrect."

It IS unfortunate you overlooked credible opinions of which you 'by and large' have concurred with. You may consider revisiting a few of the comments provided in the "optical cable vs hdmi" thread. We can add to that the three reviews that I have provided. Nothing in-credible in any of these observations...

While you may disagree with the Onkyo Reviewer's explanation of the J-test, it is the measurement and explanation of audibility that most interest me, while the significance of ommitting synthetic/stimulative data during hdmi J-test iterations doesn't seem readily apparent.

I've read that "science is not proven by democracy." However, in the absence of ABX listening tests, the balance of opinions against jitter having an audible impact in home audio equipment continues to rise. The reliability of this stance is fortified by auditory acuity knowns and measurements of jitter related distortion.
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post #22 of 284 Old 05-18-2014, 11:49 AM
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I thought jitter was a non-issue in audio for quite some time. Now, alcohol withdrawal syndrome is a different matter entirely.
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post #23 of 284 Old 05-18-2014, 02:18 PM
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Originally Posted by UndersAVS View Post

It IS unfortunate you overlooked credible opinions of which you 'by and large' have concurred with.
I have not overlooked any credible opinions. I will respond to the other quotes you provided.
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You may consider revisiting a few of the comments provided in the "optical cable vs hdmi" thread. We can add to that the three reviews that I have provided. Nothing in-credible in any of these observations...
Feel free to post them here and I will respond as time permits.
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While you may disagree with the Onkyo Reviewer's explanation of the J-test, it is the measurement and explanation of audibility that most interest me, while the significance of ommitting synthetic/stimulative data during hdmi J-test iterations doesn't seem readily apparent.
Well, let's address that now. You said:
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Originally Posted by UndersAVS View Post

http://archimago.blogspot.com/2013/10/measurements-onkyo-tx-nr1009-as-hdmi.html



The graphs above illustrate hdmi jitter under -100db. DUT is an onkyo-tx-nr1009 AV Receiver.

Your conclusion is not supported by real measurements. The measurement above is performed using a sound card and not a professional calibrated test instrument as I have been using. That aside, he has provided a tiny graph spanning large range of frequencies from 5 to 18 Khz. That masks low frequency jitter components that are bunched up on top of the main signal tone (the tall spike). Here is my measurements with appropriate level of zooming:

i-tn5rJfr-X3.png

Now you can see the sidebands clearly and they rise up to around -90 db from me eyeballing it. Sure, it is still low level but it is not under -100 db as you state.

Additionally there are those shoulders, or broadening of the signal. That is low-frequency random jitter. It is difficult to characterize what it means as far as audibility.

BTW the author of that measurement says this: "As you can see, the Onkyo is quite jittery in general whether HDMI or the other SPDIF interfaces. " He goes to show the measurements of a TEAC DAC (?) that is lower.
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I've read that "science is not proven by democracy." However, in the absence of ABX listening tests, the balance of opinions against jitter having an audible impact in home audio equipment continues to rise. The reliability of this stance is fortified by auditory acuity knowns and measurements of jitter related distortion.
There is incredible chanting about jitter in online forums and blogs but scant few people understand what it is as I showed in the example review you listed. This is not an easy concept to understand and even harder to accept. Quantity of opinion therefore does not add to one's case.

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post #24 of 284 Old 05-18-2014, 07:09 PM - Thread Starter
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Quote:
Originally Posted by amirm View Post

You said:

http://archimago.blogspot.com/2013/10/measurements-onkyo-tx-nr1009-as-hdmi.html

"The graphs above illustrate hdmi jitter under -100db. DUT is an onkyo-tx-nr1009 AV Receiver."

Your conclusion is not supported by real measurements. The measurement above is performed using a sound card and not a professional calibrated test instrument as I have been using. That aside, he has provided a tiny graph spanning large range of frequencies from 5 to 18 Khz. That masks low frequency jitter components that are bunched up on top of the main signal tone (the tall spike).

Perhaps your test instrument is more accurate on this occasion. Maybe not. There is no certainty. Certainly obvious are the noisiest sidebands @ -100db. IOW, a whisper presented simultaneously with a peak reference levels tone a few kHz away. Much lower and the distorted sound would just hit the lowest threshold of audibility with no other sound present. The visuals could be more detailed, but there really isn't much else needed than what was presented.
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Here is my measurements with appropriate level of zooming:

i-tn5rJfr-X3.png

Now you can see the sidebands clearly and they rise up to around -90 db from me eyeballing it. Sure, it is still low level but it is not under -100 db as you state. Additionally there are those shoulders, or broadening of the signal. That is low-frequency random jitter. It is difficult to characterize what it means as far as audi
bility.

Here, here now. The spec I stated was quoted from the measurement of the tx-nr1009 model. Perhaps you measured another model. Only you *might* know which model you tested. We're now back to my original question. Neg 90dB, only a whisper of audibility competing simultaneously with near reference levels.
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BTW the author of that measurement says this: "As you can see, the Onkyo is quite jittery in general whether HDMI or the other SPDIF interfaces. " He goes to show the measurements of a TEAC DAC (?) that is lower.
There is incredible chanting about jitter in online forums and blogs but scant few people understand what it is as I showed in the example review you listed. This is not an easy concept to understand and even harder to accept. Quantity of opinion therefore does not add to one's case.

Jittery in comparison, but inaudible versus less inaudible is not worth arguing, particularly when the Reviewer also comments:

"Remember that this test is synthetic and stimulative. What you see measured is not something you're probably ever going to "hear" in real music! The noise floor is not going to be down to the last bit in 16-bit audio and essentially impossible with recorded 24-bit audio (unless it's purely computer synthesized music). Also, jitter is more pronounced in the higher frequencies (11kHz and 12kHz are used as the primary signals in the J-Test). Realize that the human hearing sensitivity is well on its way down by 5kHz (as can be seen by the Fletcher-Munson Curves). Furthermore, if we specifically look at the Onkyo's J-test spectrum, the most pronounced side bands are about -90dB below the primary signal. To make matter even less worrisome is that the tall sidebands are all +/-250Hz around the primary signal and the audibility would be masked even if one did have awesome auditory acuity at 11/12kHz and could hear a signal 90dB down! This is also why I feel adding up all those sideband peaks and calling it a number (whatever picosecond or nanosecond) is really not all that useful when it comes to audibility."

Forget all of the chatter. Those of whom I've been listening seem to understand F&M curve, masking, and the relationship to room SPL.
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post #25 of 284 Old 05-18-2014, 07:58 PM
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Quote:
Originally Posted by UndersAVS View Post

"Remember that this test is synthetic and stimulative. What you see measured is not something you're probably ever going to "hear" in real music! The noise floor is not going to be down to the last bit in 16-bit audio and essentially impossible with recorded 24-bit audio (unless it's purely computer synthesized music). Also, jitter is more pronounced in the higher frequencies (11kHz and 12kHz are used as the primary signals in the J-Test). Realize that the human hearing sensitivity is well on its way down by 5kHz (as can be seen by the Fletcher-Munson Curves). Furthermore, if we specifically look at the Onkyo's J-test spectrum, the most pronounced side bands are about -90dB below the primary signal. To make matter even less worrisome is that the tall sidebands are all +/-250Hz around the primary signal and the audibility would be masked even if one did have awesome auditory acuity at 11/12kHz and could hear a signal 90dB down! This is also why I feel adding up all those sideband peaks and calling it a number (whatever picosecond or nanosecond) is really not all that useful when it comes to audibility."
Some of this is right. Some of it wrong.
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Forget all of the chatter. Those of whom I've been listening seem to understand F&M curve, masking, and the relationship to room SPL.
The forum is about chatter. If you are not interested in that or my comments about the rest of the blogs you posted, then sure, sign off and believe what you want smile.gif.

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post #26 of 284 Old 05-21-2014, 04:59 PM - Thread Starter
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Of course I'm interested in the forum and your comments. Don't let my branding of certain comments as chatter, or denunciation of others lead you to any other conclusion.

Repeating my stance regarding your previous comment:
- there is no certainty that the Reviewer's Onkyo measurements are inaccurate (not "real measurements") because a PC sound card was used
- you haven't specified which Onkyo product you measured
- can't conclude that the Reviewer's measurement are wrong (not "real measurements") because your results are different
- though the Reviewer states "a lot of jitter," the Reviewer's conclusion is that -90db distortion is completely inaudible

Am I not comprehending the significance of the difference between "low frequency jitter components that are bunched up on top of the main signal tone" which aren't easily observed in the Reviewer's graph, but apparent in your illustration?

Based upon the opinions you have given thus far, it seems that you believe engineering principles/studies cannot explain whether audibility of jitter measured -90db can be discerned, and double-blind listening tests are needed. True or false?
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post #27 of 284 Old 05-21-2014, 06:41 PM
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Quote:
Originally Posted by UndersAVS View Post

Of course I'm interested in the forum and your comments. Don't let my branding of certain comments as chatter, or denunciation of others lead you to any other conclusion.

Repeating my stance regarding your previous comment:
- there is no certainty that the Reviewer's Onkyo measurements are inaccurate (not "real measurements") because a PC sound card was used
Lack of certainty does not mean it is accurate either. It is unknow. That is the only known. smile.gif
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- you haven't specified which Onkyo product you measured
You didn't ask smile.gif. It was listed in my article where those measurements came from. It is the Onkyo TX-SR805.
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- can't conclude that the Reviewer's measurement are wrong (not "real measurements") because your results are different
The question is whether you want to really believe something or want it to be based on how we do things properly. When something is of unknown accuracy, that is what it is. You can't ask me to comment on it because it is not proper measurements. It may be accurate, it may not be. You simply don't know.
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- though the Reviewer states "a lot of jitter," the Reviewer's conclusion is that -90db distortion is completely inaudible
The comment that there is a "lot of jitter" is objective. He goes on to show another device which has less jitter. HIs comment that it is "inaudible" is an assertion. He may be right. He may be wrong. He needs to walk us through the proof point of how he arrived at that conclusion. I could tell you -40 db distortion is inaudible in a blog. Such a comment is neither here, nor there. He is posting under an alias with no qualifications in psychoacoustics. If he had such qualifications then we could respect his opinion. But we can't any more than another guy on the Internet saying smoking oregano cures lung cancer smile.gif.

Again, let me emphasize that what he says may very well be true. But no objective person would take a simple assertion as factually accurate. Per above, you just don't know. To the extent you believe it anyway, then you are going by your gut feeling and your bias in this matter. Neither way is scientific or objective way to evaluate data. An objective person at all times remains neutral. Once you are not, the rest of this discussion becomes academic because you are not interested in a discussion of science and engineering and we are into the realm of human emotions and motivations.
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Am I not comprehending the significance of the difference between "low frequency jitter components that are bunched up on top of the main signal tone" which aren't easily observed in the Reviewer's graph, but apparent in your illustration?
I don't know what you mean by "significance." If by that you mean not being able to read the true jitter components the answer is yes. His measurements are improperly done to reveal such distortions. As you can imagine, it is pretty easy to hide defects by simply "zooming out." All variations start to get smooth the more you zoom out.

Once we have the data then we can decide if it is "significant." But without it, we have no choice but to state that he is showing us measurements of jitter that hide jitter components. We have to show all the data and not censor it at the source.
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Based upon the opinions you have given thus far, it seems that you believe engineering principles/studies cannot explain whether audibility of jitter measured -90db can be discerned, and double-blind listening tests are needed. True or false?
That is false. You have not read all that I have written and have a mistaken assumption about my position. I am not repeating them here because we don't even seem to be agreeing on the data itself. For example you keep saying -90 db when I have shown you measurements with much higher levels of distortions. This is easy stuff. If we can't agree on what a chart is showing and what data is presented, we have no prayer of making progress on more complicated topic of psychoacoustics.

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post #28 of 284 Old 05-21-2014, 07:33 PM
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Originally Posted by UndersAVS View Post

Of course I'm interested in the forum and your comments. Don't let my branding of certain comments as chatter, or denunciation of others lead you to any other conclusion.

Repeating my stance regarding your previous comment:
- there is no certainty that the Reviewer's Onkyo measurements are inaccurate (not "real measurements") because a PC sound card was used

That certainty can exist if it passes additional tests. The problem isn't that it is a sound card, the problem is that its jitter performance is not known well enough. In general audio interfaces used for professional work perform well for jitter and everything else.
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- can't conclude that the Reviewer's measurement are wrong (not "real measurements") because your results are different

Agreed. There are many possible explanations for different results when so much seems to be unknown.

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- though the Reviewer states "a lot of jitter," the Reviewer's conclusion is that -90db distortion is completely inaudible

I showed how to compare jitter measurements to the thresholds of audibility for jitter as presented in standard, authoritative texts in the "Coax versus HDMI" thread that was closed.
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I'm I not comprehending the significance of the difference between "low frequency jitter components that are bunched up on top of the main signal tone" which aren't easily observed in the Reviewer's graph, but apparent in your illustration?

The audibility of jitter varies with the modulating frequency, as I showed in the now closed "Coax versus HDMI" thread. I also showed how to adjust measurements that are made with different test tones.
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Based upon the opinions you have given thus far, it seems that you believe engineering principles/studies cannot explain whether audibility of jitter measured -90db can be discerned, and double-blind listening tests are needed. True or false?

Well there's one little problem, and that is that analysis based on standard, authoritative texts about the audibility of jitter says that these quantities and and modulating frequencies fall well short of producing audible jitter.

One reason that HDMI jitter tends to be numerically greater is that HDMI audio is transmitted in widely separated possibly varying and therefore somewhat intermittent fairly long packets, which makes the job of the buffer on the receiving end that much harder. In a SP/DIF connection, the audio information is transmitted in small packets that follow each other closely with consistent timiing.

Therefore HDMI audio has to be buffered and reclocked no matter what format it is in.

http://en.wikipedia.org/wiki/HDMI#Audio.2Fvideo

"
TMDS
Transition Minimized Differential Signaling (TMDS) on HDMI interleaves video, audio and auxiliary data using three different packet types, called the Video Data Period, the Data Island Period and the Control Period.[64] During the Video Data Period, the pixels of an active video line are transmitted.[64] During the Data Island period (which occurs during the horizontal and vertical blanking intervals), audio and auxiliary data are transmitted within a series of packets.[64] The Control Period occurs between Video and Data Island periods.[64]

Both HDMI and DVI use TMDS to send 10-bit characters that are encoded using 8b/10b encoding that differs from the original IBM form for the Video Data Period and 2b/10b encoding for the Control Period. HDMI adds the ability to send audio and auxiliary data using 4b/10b encoding for the Data Island Period.[64] Each Data Island Period is 32 pixels in size and contains a 32-bit Packet Header, which includes 8 bits of BCH ECC parity data for error correction and describes the contents of the packet.[65] Each Packet contains four subpackets, and each subpacket is 64 bits in size, including 8 bits of BCH ECC parity data, allowing for each Packet to carry up to 224 bits of audio data.[66] Each Data Island Period can contain up to 18 Packets.[67] Seven of the 15 Packet types described in the HDMI 1.3a specifications deal with audio data, while the other 8 types deal with auxiliary data.[65] Among these are the General Control Packet and the Gamut Metadata Packet. The General Control Packet carries information on AVMUTE (which mutes the audio during changes that may cause audio noise) and Color Depth (which sends the bit depth of the current video stream and is required for deep color).[68][69] The Gamut Metadata Packet carries information on the color space being used for the current video stream and is required for xvYCC.[53][70][71]
"

I've spent my life reading documents like this, and IMO its difficulty is about 90th percentile.

This article provides more information and some pictures which may help:

http://www.wireworldcable.com/hdmi_tech.html

"HDMI® employs Transition Minimized Differential Signaling (TMDS) transmitted over 4 pairs of wires to carry video, audio and auxiliary data via one of three modes, called the Video Data Period, the Data Island Period and the Control Period. During the Video Data Period, the pixels of an active video line are transmitted. During the Data Island period (which occurs during the horizontal and vertical blanking intervals), audio and auxiliary data are transmitted within a series of packets. The Control Period occurs between Video and Data Island periods."




Let's start out relatively simple - with S/PDIF. S/PDIF data streams are composed of data blocks:

Reference: http://www.nutsvolts.com/uploads/magazine_downloads/873/HDTV.pdf

In S/PDIF Audio is not transmitted as a uniform continuous stream of 16 bit samples. Instead, audio data regardless of sample format is padded out into 32 bit subframes. There are 2 subframes (1 per channel) per frame. Frames are collected into groups of 192 which are called data blocks. Multichannel recordings are usually handled internally as multiple stereo streams. This frame structure creates an opportunity for jitter because there is a break in the data every 192 sample pairs. With 44.1 KHz sampling this works out to be about 230 Hz (or 250 Hz for 48 KHz).

Sometimes jitter artifacts at multiples of 230 or 250 Hz can be observed in the outputs of DACs. There is also a data structure on the CD which groups samples for the purpose of error detection and correction. This provides another opportunity for poorly designed circuitry to create jitter artifacts at other frequencies in the same general range.

So much for S/PDIF which isn't all that simple! ;-) Now lets talk about HDMI.

One of the surprises about HDMI is that its data format hasn't changed that much from the days of CRT displays. The data for each scan line is grouped together. Near the end of the scan line there is an allowance for what used to be called the retrace interval, which was required in the days of analog video processing and CRT displays to allow CRT beams to skip back to the left edge of the screen. During the retrace interval the display device blacked out or blanked the screen, so this time period could be used to hold any non-video (or in HDMI terminology Auxiliary Data which includes sound) data that needs to be transmitted. In the days of NTSC video these periods of blanked out video were used for things like closed captioning.

There are both horizontal retrace intervals which are relatively short and frequent, and vertical retrace intervals which are usually longer and far less frequent. Therefore sound data is showing at the front door of the audio circuitry in an AVR in little batches that are interrupted at both the horizontal scanning rate (one cycle for every horizontal row of pixels) and the vertical scanning rate (one cycle for every video frame or still picture that is transmitted in quick succession to create the impression of motion video.

So, poorly designed equipment can add jitter at both the overall frame rate (24, 30, or 60 Hz) and the horizontal scan rate - 480, 720, or 1080 lines per screen full of data for the most common video standards.

I've been sitting here watching people pontificate about jitter, knowing from my own studies and lab work that certain jitter frequencies will pop out over and over again due to the fact that audio data is transmitted mixed in with video data, but in various sized groups. AFAIK nobody said nuttin', and that gives me an idea about who is posing and who actually plays. ;-)

People who understand what they need to about HDMI may realize that HDMI jitter frequencies are often keyed to video format because the audio data is mixed in with the video. Therefore, analysis of audio measurements have to take picture format into account, which unfortunately one rarely if ever sees in published test reports.
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post #29 of 284 Old 05-22-2014, 10:01 AM
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Why does the WireWorld example show color burst? There is no color burst on component video and HDMI does not support old composite digital video. rolleyes.gif

But then what more should we expect from yet another audiophool company - WireWorld!

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post #30 of 284 Old 05-22-2014, 01:19 PM
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Why does the WireWorld example show color burst? There is no color burst on component video and HDMI does not support old composite digital video. rolleyes.gif

But then what more should we expect from yet another audiophool company - WireWorld!

Good point. I updated the diagram around 4:18 EDT.
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