I am 100,000,000,000 % with you on this Ron. To me the best asset a music lover can have is the ability to swith the critical mode of thinking off at will, anytime, anywhere. While politically I am a centrist, when it comes to all things audio I'm a dyed in the wall liberal.
Jack, this for me is a key criterion for a system on song. When not quite on the money then, yes, you have to quite deliberately switch off your critical faculties, because it is easy to hear where the system is not quite right. But when it passes through that crucial barrier of performance that effort on your part is no longer needed, you can just allow yourself to be swept away with the emotion of the musical event. And there are no nasty surprises: when the soprano hits that big, big note and holds it, you don't wince internally, because as far as the system is concerned it's effortless, it just flows out with the ease that it would had you experienced that person singing in the flesh ...
Jack, this for me is a key criterion for a system on song. When not quite on the money then, yes, you have to quite deliberately switch off your critical faculties, because it is easy to hear where the system is not quite right. But when it passes through that crucial barrier of performance that effort on your part is no longer needed, you can just allow yourself to be swept away with the emotion of the musical event. And there are no nasty surprises: when the soprano hits that big, big note and holds it, you don't wince internally, because as far as the system is concerned it's effortless, it just flows out with the ease that it would had you experienced that person singing in the flesh ...
This is the point, of course, Jack, when the welding of the power chord of the food processor to the speaker terminals of the HTIB frees the audio artificial intelligence of the system created by the defeating of all cell phones in the area code, and the system begins to understand the artist's intention better than himself, pushing forward the good notes and squishing the bad ones into the blu-tak coupling the speakers to the faux marble statue of the Virgin Mary. Have real marble? The soundstage will reach from Australia to Brazil!
Jack, this for me is a key criterion for a system on song. When not quite on the money then, yes, you have to quite deliberately switch off your critical faculties, because it is easy to hear where the system is not quite right. But when it passes through that crucial barrier of performance that effort on your part is no longer needed, you can just allow yourself to be swept away with the emotion of the musical event. And there are no nasty surprises: when the soprano hits that big, big note and holds it, you don't wince internally, because as far as the system is concerned it's effortless, it just flows out with the ease that it would had you experienced that person singing in the flesh ...
You have added nothing new to the discussion, notwithstanding my inquiries (and those of Tim). Instead you repeat the essence of subjectivity but state it in such a manner that you believe everyone will, or at least should, concur in *your* personal assessment.
Just re-read your own language. Sheesh. You speak about everyone when all you're really talking about is just you. "You have to quite deliberately switch off your critical faculties". Really?!? *I* do? How the heck do you know what I *have* to do. Sometimes an ounce of humility goes a long way. You would be well served to recognize that, just like every living creature on this planet, you have boundaries.
If you have nothing new to offer up to members, then I will presume to tell you on behalf of the entire community here that we all understand your point of view but it grows tiresome to re-read it ad nauseum.
I'm sorry I have provoked a strong response from you, Ron, but I feel you overreacting to my use of "you". Here I was responding to Jack's comment, so in that sense the "you" was Jack, not everyone that happened to be reading, it was a "private" acknowledgement of what frequently occurs when listening to systems. Perhaps in context it would be construed as signifying everyone that reads my post, but in my defence I would put forward the constant posting of comments by members along the lines of "but of course it never sounds real, it always sounds like a hifi system, it can never do otherwise". This seems to be a generally accepted viewpoint, repeated ad nauseum, so I think I have pretty good grounds for suggesting that the majority of listeners do have to "quite deliberately switch off their critical faculties".
And I thought that the point of this forum is What's Best. So is the reality that we accept that achieving a high level of reproduction is "intrinsically" doomed to failure, and so all we can hope to do otherwise is layer significant wads of money around our gear in the hope that this somehow alleviates the disappointment of not achieving high standards of replay by just pushing the right buttons? In essence, I'm saying there is Another Way, and I apologise if this disturbs people ...
And I thought that the point of this forum is What's Best. So is the reality that we accept that achieving a high level of reproduction is "intrinsically" doomed to failure, and so all we merely hope to do otherwise is layer significant wads of money around our gear in the hope that this somehow alleviates the disappointment of not achieving high standards of replay by just pushing the right buttons? In essence, I'm saying there is Another Way, and I apologise if this disturbs people ...
No apology needed Frank. There are many who claim we should have declared victory and gone home long ago, They bare an amazing resemblance to those who lament the inherent imperfection of audio reproduction. Don't be afraid to take time out chase butterflies once and awhile. Those who don't like itknow what they can do. (Push the ignore button)
No need to apologize to the whole WBF community - at less to me, I do not speak for others. Your direct and enthusiast style using "you" is normal in friendly discussions, and although I am a non native english reader I could easily understand it.
BTW, it comes to me as a sad surprise that someone in WBF writes "I will presume to tell you on behalf of the entire community here that we all understand your point of view but it grows tiresome to re-read it ad nauseum". As far as I know we have not signed any endorsement of opinions or nominated representatives when joining WBF.
In standard consumer systems, the source is the master. This means it that it determines the pace with which the DAC consumes the data. Worse yet, the data does not arrive with a timestamp saying "play this then" but rather, the rate of delivery of the digital samples determines that. The DAC then has to sample that analog timing and track its rate. This results in increased jitter.
So we introduce jitter in the system when we didn't need to. We call this type of architecture "push" system where the source is pushing data out. The other model is a pull system where the target (or the "sink") asks for data at the rate it wants to play. Had we done that, then jitter upstream of the target would not have mattered.
This is probably an artifact of older architecture designed for cost as 30 years ago, memory was expensive and having the pull method means reading and storing data until you need it. The amount that needs to be stored can be potentially large if one accommodates a situation where the CD drive would need to seek and read data from a slow optical drive. Today this kind of memory cost next to nothing. But we continue to build devices the old fashion way.
Now, there are ways around this and computer based systems can approximate it (i.e. using asynchronous USB bridges) or outright implement it in the form of a dedicated player. PS Audio type devices are an example.
Alas, HDMI follows the same path. In its default mode it runs using the same push method. It does have a provision for the target to set the data rate though but it is rarely implemented and even there, one doesn't know how well it works.
But do we have a "real" problem then? If the "normal" consumer is happy with what he gets, and the fussier one can achieve "ideal" results by using one of the alternative solutions, is there really a problem? Even the HDMI can be "fixed" by setting up a big enough buffer, which as you say, is cheap as chips. So is anything really "broken"?
But do we have a "real" problem then? If the "normal" consumer is happy with what he gets, and the fussier one can achieve "ideal" results by using one of the alternative solutions, is there really a problem?
Yes because it is very easy to have (relatively speaking) high levels of distortion here and it is something that is never spec'ed and worse yet, is system interdependent.
Even the HDMI can be "fixed" by setting up a big enough buffer, which as you say, is cheap as chips. So is anything really "broken"?
No, buffering never fixes what I explained. You can buffer all you want. At playback time, that buffer will be drained based on the clock that was inputting to that buffer, not output.
The buffer I spoke of was in the case where the system function could be reversed. Then it serves a useful mode. But in our current systems, it solves nothing regarding fidelity issues.
I listened to tracks 13 and 14(both all digital) and compared them to track 12 which is analog. Damn good recordings. Track 14 is pretty fantastic, generally I can tell the size of the venue as there is a ambient halo around the sound. Tracks 13 and 14 are the U.S. Naval Academy Band and the perspective seems smaller then the analog,that is until I stood up and finally got a true sense of space. I prefer the analog, it actually sounds less sterile but the digital is very dynamic. These digital recordings are very good because with tape the soundstage is always more open,but hearing the Naval Academy Band I got the sense of standing in the middle of a vast soundfield, Bravo.
12. Strauss: Egyptian March
13. Ohlemeyer: Sesquicentennial March
14. Hosay: Since We First Set Sail
p.s. I need to point out that my system is designed to recover as much ambient information that is possible. That is why tape has always sounded better to me. Most digital recordings that I have seem closed in and are lacking the openness of analog especially tape. When I play a exceptional recording I can stand in the middle at 5 ft in front of the speakers and be enveloped in a vast sound field. Even 30 feet away the ambient signature makes the recording extremely lifelike.
Like I said tracks 13 and 14 seemed to have bridged the gap (95 pct) in my system.
No, buffering never fixes what I explained. You can buffer all you want. At playback time, that buffer will be drained based on the clock that was inputting to that buffer, not output.
The buffer I spoke of was in the case where the system function could be reversed. Then it serves a useful mode. But in our current systems, it solves nothing regarding fidelity issues.
But a buffer can be set up, quite easily in fact, which is driven by 2 clocks. The input clock fills the buffer at its rate, and the output one empties it at its very, very precise, jitter free rate. Now, typically there will be a discrepancy between the average rate of these 2 clocks, meaning the buffer will slowly overfill, or be starved. But if the buffer is big enough, and there is intelligent synchronising between the processes, then it will work perfectly well. The Genesis Digital Lens, about 20 or so years ago used this technique.
But a buffer can be set up, quite easily in fact, which is driven by 2 clocks. The input clock fills the buffer at its rate, and the output one empties it at its very, very precise, jitter free rate. Now, typically there will be a discrepancy between the average rate of these 2 clocks, meaning the buffer will slowly overfill, or be starved. But if the buffer is big enough, and there is intelligent synchronising between the processes, then it will work perfectly well. The Genesis Digital Lens, about 20 or so years ago used this technique.
Such a scheme will cause lip sync to be lost in video cases since that drift, is clock sync being lost.
For audio only, it also breaks. Imagine if I feed that DAC my satellite audio. I leave that box on 24 hours a day. What type of buffer you need for months of overrun? Your DAC doesn't know who is feeding it data.
And how do you deal with underrun? Pause and glitch? Or Resample? Both are consequences of a system not working as specified.
I have only seen one system do as you say and it was a DAC. It assumed audio-only solution and had a detection mechanism to guess the clock speed of the source and then use one of a number of frequencies. Seemed like a clever kludge but again, it would break if fed anything that was not pure audio source.
SAS/SATA controllers include an elasticity buffer to deal with the synch issue, and have mechanisms defined to maintain synchronization over time (SKP cycles). Is there anything similar in the digital audio/video world? I do not know, am curious. - Don
p.s. I must note that these schemes, properly implemented, still drop a data frame now and then to keep things in synch. I do not know how audible/visible such artifacts would be; certainly aan issue if an entire frame is lost. Seems like a small packet size would be undetectable, but of course harder to implement.
Damn good recordings. Track 14 is pretty fantastic, generally I can tell the size of the venue as there is a ambient halo around the sound. Tracks 13 and 14 are the U.S. Naval Academy Band and the perspective seems smaller then the analog,that is until I stood up and finally got a true sense of space. I prefer the analog, it actually sounds less sterile but the digital is very dynamic. These digital recordings are very good because with tape the soundstage is always more open,but hearing the Naval Academy Band I got the sense of standing in the middle of a vast soundfield, Bravo.
12. Strauss: Egyptian March
13. Ohlemeyer: Sesquicentennial March
14. Hosay: Since We First Set Sail
Good to hear, Roger: it's interesting that the soundstage is more open with the tape, it shouldn't be that way; perhaps the somewhat "warmer" sound of the tape contributes to that sense.
But maybe the clue is when you say the analogue masters are less sterile, but the digital originals are dynamic. I've experienced this effect many times, and have always found it to be that the dynamics of the true digital are actually stressing the overall capabilities of the playback somewhat more, the system has a harder time rendering the sound without audible distortion. And that distortion comes across, creates the subjective feeling, that the sound is "sterile", lacking musicality.
A good analogy is a comfortable old family sedan, which cruises very nicely at 70mph. But if it's asked to pick up the pace and run continuously at 85mph everything feels less relaxed, it's no longer in its comfort zone, and it's just not a pleasant experience to be in the car. Unless you swap to a vehicle that very specifically has been designed to handle the greater pace, you'll always prefer the slower running rate ...
But if the whole digital stream, HDMI say, is buffered then the video and audio always remain in alignment. Plenty of memory, but that's cheap ...
For audio only, it also breaks. Imagine if I feed that DAC my satellite audio. I leave that box on 24 hours a day. What type of buffer you need for months of overrun? Your DAC doesn't know who is feeding it data.
You can attach a reasonable amount of intelligence to the synchronising mechanism: it waits until there is an audible silence of a few samples, and adjusts the buffer pointers. Same sort of principle is used in digital volume controls. Essentially, if there is a need to realign the data streams, smart processing will pick a moment when it will have minimal audible effect.
And how do you deal with underrun? Pause and glitch? Or Resample? Both are consequences of a system not working as specified.
You only need to straighten out the timing at the last moment, just before the DAC. All the various digital funnies are handled in the realm of the jittery input
I have only seen one system do as you say and it was a DAC. It assumed audio-only solution and had a detection mechanism to guess the clock speed of the source and then use one of a number of frequencies. Seemed like a clever kludge but again, it would break if fed anything that was not pure audio source.
SAS/SATA controllers include an elasticity buffer to deal with the synch issue, and have mechanisms defined to maintain synchronization over time (SKP cycles). Is there anything similar in the digital audio/video world? I do not know, am curious. - Don
Oh, there are plenty of interfaces with flow control. But this one is not it as there is no return path. HDMI does have a return path but as I noted, that scheme is not used much today.
p.s. I must note that these schemes, properly implemented, still drop a data frame now and then to keep things in synch. I do not know how audible/visible such artifacts would be; certainly aan issue if an entire frame is lost. Seems like a small packet size would be undetectable, but of course harder to implement.
No, that is not allowed. When we stream video on the Internet or when using software players, that can happen as the systems are not real time. But in hardware devices that cannot happen or the device is broken.
I can't speak for video. I'm sure that on some level, my video loses sync and lips don't align perfectly with speech for a moment. Sometimes I even notice. But when it comes to audio, I really pay attention, and I don't hear jitter. I don't doubt that some, trained to listen for some artifact or another, do, but not most of us. Not even most audiophiles with high-dollar, high-resolution systems. Google "audibility of jitter." Look past all the audiophiles wringing their hands over it the way we wrung them over THD two zeros to the right of the decimal point in the 70s. Look at Google Scholar. You can't get to most of it without paying a fee, which gives you an idea of its value compared to the stuff on audio asylum and hydrogen audio, but you can see that there are pages and pages of scholarly studies on the audibility of jitter. Pages of studies by smart people with access to real research labs, using controlled methodologies, trying to decide if people can hear the stuff.
It's a largely theoretical problem. Which is not to say, for the record, that it does not exist or is not, on any level, audible. But the overwhelming majority of music lovers, even the overwhelming majority of audiophiles, have much more audible fish to fry. If this one keeps you up at night, or takes any of your money this side of your absolute personal pinnacle of speakers, DAC, amp, music collection...you're worrying about the wrong stuff. MHO. YMMV.
, does anyone else have interest to know why digital audio is broken or wants to share an opinion?
