Friday, July 21, 2017
Thursday, July 20, 2017
Are Synchronous DAC receivers reappearing?
Truth be told, of course, synchronous DAC receivers had never gone away. While some of the earlier ones are discontinued, the DIR 9001 continues, and is often the receiver chosen by DIY'ers, I have noticed. I'm talking SPDIF/AES only, as you may already know I despise everything else for practical/personal reasons, AND that's a pretty cut and dried case about which I need not comment anyway.
But from barely respectable on up in manufactured gear, asynchronous receivers have/had become the norm. Few except for cranks were raising the old "puts your jitter into the data" arguments. Some equipment was giving you a choice--that's fine. But generally since the asynchronous receivers were showing better measured performance, anyone who cared was using them. Except for the universe of contrarians.
I'm using one right now with my DVD-5000 dac, which has CS8414 and dual differential 1704's. CS8414 might not have the best self-jitter, DIR 9001 may be better in that regards.
But anywayz it seems to me that synchronous receivers are required for things like HDCD, aren't they? You can get away with any sort of interpolation with that, I would think, there's meanings to certain exact sets of bits.
Then, I also think about MQA. Once again, it seems if you are encoding any extra information into the audio, that isn't the kind of thing which may be interpolated or whatever.
So I noticed in review of the latest and greatest Meridian DAC, which is wonderful for sure, mention is made of a FIFO buffer which gets jitter to below 0.5 Hz.
OK, that sounds like a synchronous receiver with a 1 second buffer, though I could be wrong about the buffer size.
Most of the receiver chips get away with what seem like tiny buffers, then often spec jitter suppression only above 20kHz. That is surely wrong, it should be at least spec'd down to 1Hz. With a long enough buffer, the DAC clock can vary slowly yet stay in line with whatever the source does, and therefore only subsonic jitter remains. I understand that people are quite sensitive in the usually subsonic 3-10 Hz range to FM, aka jitter or wow, so you have got to get below that.
Now I really do wonder what is going on inside the Denon DVD-9000 with it's 330 msec latency compared to other DACs. Is it an anti-jitter mechanism? How well does that work?
I know there's no question I have to get beyond my very limited so far jitter measurements and do a better investigation of all this, with J test and so on.
BTW, the J-test harmonics appear in most cases to be way way below -110dB, often in the -130dB range. That's a worst case jitter situation, which maybe occurs for a few seconds in a lifetime of playing. Mostly, the jitter sidebands must be way below that. And that's so negligible it's a wonder we even think about it. (Well, that's a long story of course. And it also relates to the lack of controlled blind testing. And people don't want to write off uncontrolled impressions they have had. And with digital transmission there can only be two things, data and time. And the data can be checked and shown to be perfect. So the only thing left is time. SO time MUST explain all!!!)
This seems way below the importance of more easily measured things, like harmonic (and therefore IM) distortion, which often reaches high in audiophile designs. Often distortion sidebands reach -60dB or higher, possibly 3,000 times or more larger than the sidebands being caused by jitter even through toslink, etc.
But from barely respectable on up in manufactured gear, asynchronous receivers have/had become the norm. Few except for cranks were raising the old "puts your jitter into the data" arguments. Some equipment was giving you a choice--that's fine. But generally since the asynchronous receivers were showing better measured performance, anyone who cared was using them. Except for the universe of contrarians.
I'm using one right now with my DVD-5000 dac, which has CS8414 and dual differential 1704's. CS8414 might not have the best self-jitter, DIR 9001 may be better in that regards.
But anywayz it seems to me that synchronous receivers are required for things like HDCD, aren't they? You can get away with any sort of interpolation with that, I would think, there's meanings to certain exact sets of bits.
Then, I also think about MQA. Once again, it seems if you are encoding any extra information into the audio, that isn't the kind of thing which may be interpolated or whatever.
So I noticed in review of the latest and greatest Meridian DAC, which is wonderful for sure, mention is made of a FIFO buffer which gets jitter to below 0.5 Hz.
OK, that sounds like a synchronous receiver with a 1 second buffer, though I could be wrong about the buffer size.
Most of the receiver chips get away with what seem like tiny buffers, then often spec jitter suppression only above 20kHz. That is surely wrong, it should be at least spec'd down to 1Hz. With a long enough buffer, the DAC clock can vary slowly yet stay in line with whatever the source does, and therefore only subsonic jitter remains. I understand that people are quite sensitive in the usually subsonic 3-10 Hz range to FM, aka jitter or wow, so you have got to get below that.
Now I really do wonder what is going on inside the Denon DVD-9000 with it's 330 msec latency compared to other DACs. Is it an anti-jitter mechanism? How well does that work?
I know there's no question I have to get beyond my very limited so far jitter measurements and do a better investigation of all this, with J test and so on.
BTW, the J-test harmonics appear in most cases to be way way below -110dB, often in the -130dB range. That's a worst case jitter situation, which maybe occurs for a few seconds in a lifetime of playing. Mostly, the jitter sidebands must be way below that. And that's so negligible it's a wonder we even think about it. (Well, that's a long story of course. And it also relates to the lack of controlled blind testing. And people don't want to write off uncontrolled impressions they have had. And with digital transmission there can only be two things, data and time. And the data can be checked and shown to be perfect. So the only thing left is time. SO time MUST explain all!!!)
This seems way below the importance of more easily measured things, like harmonic (and therefore IM) distortion, which often reaches high in audiophile designs. Often distortion sidebands reach -60dB or higher, possibly 3,000 times or more larger than the sidebands being caused by jitter even through toslink, etc.
DAC low level pictures
Marvey at SuperBestAudioFriends has posted pictures of low level resolution of 3 DAC's, the Schiit Yggdrasil, the MSB analog DAC (which has long been on my want list), and the Nad M51.
At -90.31 dBFS (basically near the limits of 16 bit audio) the Schiit is showing a trifle of notchiness around the zero crossing point, otherwise a fairly smooth recongnizable sine wave.
The MSB is showing notching all over the wave. (And Marvey comments that John Atkinson didn't give this DAC any crap for only achieving 18 bits resolution, etc.) and looks a lot worse.
The NAD M51 (which uses a very high frequency PWM conversion--the very kind I find the least intellectually acceptable) is showing an almost perfect looking sine wave, just a slight bit of lumpiness at the extremes. This may be the best looking sine I've ever seen at -90dBFS.
At -90.31 dBFS (basically near the limits of 16 bit audio) the Schiit is showing a trifle of notchiness around the zero crossing point, otherwise a fairly smooth recongnizable sine wave.
The MSB is showing notching all over the wave. (And Marvey comments that John Atkinson didn't give this DAC any crap for only achieving 18 bits resolution, etc.) and looks a lot worse.
The NAD M51 (which uses a very high frequency PWM conversion--the very kind I find the least intellectually acceptable) is showing an almost perfect looking sine wave, just a slight bit of lumpiness at the extremes. This may be the best looking sine I've ever seen at -90dBFS.
Sigma Delta "Impulsivity"
One thing I've noticed recently in both 1-bit and multibit sigma delta DACs is something I call "impulsivity." Peaks of loudness seem to get peakier, like tiny highlight spots of glare.
Now, firstly this could be just my imagination, I'm not going to even try to prove otherwise.
Secondly, sigma delta DACs, at least the better ones for quite some time have had better S/N and dynamic range specs than even the best, yes even my sacred 1704's. So you could really say, black is blacker and therefore things rise up from the black with more contrast. More contrast means more "peakier."
I'm suggesting something different. I'm suggesting that when the sigma delta DAC has to output it peak it has to "work harder." The way sigma delta dacs work is by a feedback loop, which drives the narrow converter to push the output into the correct signal. When there is an actual peak in the signal, the overload is higher and the feedback has to work "harder," or at least more consistently, to one direction. This is precisely the kind of thing our neural networks are designed to detect: correlation and causality and therefore intensity of effort. We feel when things are struggling, or just loafing along.
The PCM dac, on the other hand, never "struggles." It has the large jumps pre-fabricated and ready to go when needed. It simply assembles the pieces for any given sample, and that is that. It is therefore imperturbable (just loafing along).
Now I'm not suggesting some sort of ESP being involved, so how could we sense such things? The answer is: it's below the noise floor, and the ultimate timing of things.
Now, firstly this could be just my imagination, I'm not going to even try to prove otherwise.
Secondly, sigma delta DACs, at least the better ones for quite some time have had better S/N and dynamic range specs than even the best, yes even my sacred 1704's. So you could really say, black is blacker and therefore things rise up from the black with more contrast. More contrast means more "peakier."
I'm suggesting something different. I'm suggesting that when the sigma delta DAC has to output it peak it has to "work harder." The way sigma delta dacs work is by a feedback loop, which drives the narrow converter to push the output into the correct signal. When there is an actual peak in the signal, the overload is higher and the feedback has to work "harder," or at least more consistently, to one direction. This is precisely the kind of thing our neural networks are designed to detect: correlation and causality and therefore intensity of effort. We feel when things are struggling, or just loafing along.
The PCM dac, on the other hand, never "struggles." It has the large jumps pre-fabricated and ready to go when needed. It simply assembles the pieces for any given sample, and that is that. It is therefore imperturbable (just loafing along).
Now I'm not suggesting some sort of ESP being involved, so how could we sense such things? The answer is: it's below the noise floor, and the ultimate timing of things.
Sunday, July 16, 2017
Time Alignment using Tact
It's pretty easy to set the supertweeter time alignment using the Tact as measurement tool, using the mid way Behringer DEQ to set the delay because the supertweeters are about 6 inches back. The Tact stimulus is a high frequency rich "snap", and it's easy to see where the high frequency wiggles of the supertweeter line up with the initial impulse in the panels. The initial impulse in the panels appears to go down in the Tact display apparently because of some weirdness in the Tact stimulus and display. I use SpeakerPop to set polarity.
I've determined the correct setting to require 0.2 ms delay in the panels. However, since Tact uses a 48kHz signal, which is delaying the panel approximately an additional 0.35 ms due to the DVD-5000, the total delay is about 0.55 ms, or about what it physically looks to be. I actually set the relative delay at 0.55 because I am "optimizing" for 96kHz. If I am seriously listening to 44.1 I can subtract 0.35 ms delay in the mid way DEQ to compensate for the added latency in the DVD-5000 at 44.1.
Setting the time alignment for the subs is as clear as mud. The Tact stimulus doesn't produce very much bass energy, and if you play the system normally the "bass" is just endless LF ringing.
What I have historically done, only it isn't working as good anymore (and never worked great) is separate the channels. To set right time alignment, I disconnect the left panel and sub, reverse the panel channels. Then both Tact channels play the right side, with the left panel channel actually playing the right channel. This gives a display with sub in one channel, and panel in the other channel, and you can sort of see how they line up.
But it's pretty hard to tell, because the sub is only making very small and slow waves in the Tact display, and it's not always exactly clear where the "beginning" is.
The Tact doesn't make this any easier. It does not save the previous measurement(s) so you can see how things changed. Because the snap stimulus has very little low frequency information, the subs are barely even audible. This lack of information means that each plot is going to differ from the previous one, randomly, even if you didn't make any changes (unless you do endless averaging, I suppose, I do only 10 trial averaging which already requires long waits).
Anyway, after hours of futzing with this anyway, I finally came to the idea that about 5.4 ms was the correct panel delay time align the subs. (Because the subs are further back, the panels need to be delayed to compensate, by 5.4 ms?) This makes no sense because the panels are not that far back. I made that judgement on the left side, for which the subs may be as much as 3 feet back (they're only as much as 28 inches on the other side).
Anyway, on the other side, things were even less clear. The number could have been as low as 2 ms (about what it looks) or as high as 5 ms.
Finally I came up with a clever trick in programming the DEQ's. I indeed do delay the subs (even though they are the farthest back) by 4ms. Then I delay the panels by 7.5ms and the super tweeters by 6.95 ms and I can leave those two alone. Then, I can adjust the sub delay simply by turning one knob.
And this control is very nice. Now it is very clear how the sub delay affects the sound, and exactly the opposite as I had thought. At the "minimum" measured sub delay of 3.5 ms the sound is a trifle boomy. If I dial the sub delay more, it tightens up. I had figured you always want the panels to "lead" and therefore provide the initial more perfect sound. But it appears what is really important is to not have the subs lag too far behind.
Anyway, now I have a control and I can just keep tuning by ear. By turning the sub delay up to 7.5 I could in effect make the relative delay 0, or by turning the sub delay down to 0 I can make the relative delay 7.5. So I have all the range I need in one control.
(Sadly there appears to be no way to add relative delay to the subs.)
With this control, so far I have gravitated to 3.6 ms, which would mean the panels are delayed 7.5-3.6 or 3.9 ms relative to the subs (maybe some different numbers would make this easier). I had previously been using a 3.3 ms delay as handed down from endless re-thinkings mainly.
I went ahead and put the panels at 10 ms delay. That puts the super tweeters at 9.46 ms (only 0.02 adjustments available) and the subs at 6.10 ms for the same relative delays as above, and gives me full adjustability down to panel delay of up to 10 ms or down to relative negatives just by turning the sub delay knob which turns out to have a very interesting subjective effect.
Then I discovered that running the Tact test at precisely 6.25ms (relative to 10.00 ms in the panels) show the right channel going down (the sub leading) over 100's of MS, and the left channel going up (the sub lagging). This has to be the perfect in-between compromise. At 6.00 ms and 6.5 ms both channels go either up (one more than the other) or down. So I've settled on 6.25 as my best guestimate (equal to 3.75 ms difference from the panels which are delayed by 10 ms for convenience now).
When I play 96kHz, I merely dial up the mid way delay from 10.00 (calibrated at Tact's 48 khz) to 10.38. To play 44.1kHz I should dial down the mid way delay to 9.94 (if I care). To change the bass alignment I can just adjust the bass.
Actually I'm finding that 6.50 msec delay, relative to 10.38 (for 96kHz sampling).or whatever I've set the panels to, works better than 6.25. 6.50 subjectively lets the bass be bass, to hang a bit. 6.25, what I called the "compromise" position, is too dry, the reverse angle taken by the two sides cancel or something and it sounds dry.
Each time I do these alignments via Tact some trick like this arises and gives me an "angle" to make an objective/subjective judgement. Who knows I could be off by more than 1 ms.
I've determined the correct setting to require 0.2 ms delay in the panels. However, since Tact uses a 48kHz signal, which is delaying the panel approximately an additional 0.35 ms due to the DVD-5000, the total delay is about 0.55 ms, or about what it physically looks to be. I actually set the relative delay at 0.55 because I am "optimizing" for 96kHz. If I am seriously listening to 44.1 I can subtract 0.35 ms delay in the mid way DEQ to compensate for the added latency in the DVD-5000 at 44.1.
Setting the time alignment for the subs is as clear as mud. The Tact stimulus doesn't produce very much bass energy, and if you play the system normally the "bass" is just endless LF ringing.
What I have historically done, only it isn't working as good anymore (and never worked great) is separate the channels. To set right time alignment, I disconnect the left panel and sub, reverse the panel channels. Then both Tact channels play the right side, with the left panel channel actually playing the right channel. This gives a display with sub in one channel, and panel in the other channel, and you can sort of see how they line up.
But it's pretty hard to tell, because the sub is only making very small and slow waves in the Tact display, and it's not always exactly clear where the "beginning" is.
The Tact doesn't make this any easier. It does not save the previous measurement(s) so you can see how things changed. Because the snap stimulus has very little low frequency information, the subs are barely even audible. This lack of information means that each plot is going to differ from the previous one, randomly, even if you didn't make any changes (unless you do endless averaging, I suppose, I do only 10 trial averaging which already requires long waits).
Anyway, after hours of futzing with this anyway, I finally came to the idea that about 5.4 ms was the correct panel delay time align the subs. (Because the subs are further back, the panels need to be delayed to compensate, by 5.4 ms?) This makes no sense because the panels are not that far back. I made that judgement on the left side, for which the subs may be as much as 3 feet back (they're only as much as 28 inches on the other side).
Anyway, on the other side, things were even less clear. The number could have been as low as 2 ms (about what it looks) or as high as 5 ms.
Finally I came up with a clever trick in programming the DEQ's. I indeed do delay the subs (even though they are the farthest back) by 4ms. Then I delay the panels by 7.5ms and the super tweeters by 6.95 ms and I can leave those two alone. Then, I can adjust the sub delay simply by turning one knob.
And this control is very nice. Now it is very clear how the sub delay affects the sound, and exactly the opposite as I had thought. At the "minimum" measured sub delay of 3.5 ms the sound is a trifle boomy. If I dial the sub delay more, it tightens up. I had figured you always want the panels to "lead" and therefore provide the initial more perfect sound. But it appears what is really important is to not have the subs lag too far behind.
Anyway, now I have a control and I can just keep tuning by ear. By turning the sub delay up to 7.5 I could in effect make the relative delay 0, or by turning the sub delay down to 0 I can make the relative delay 7.5. So I have all the range I need in one control.
(Sadly there appears to be no way to add relative delay to the subs.)
With this control, so far I have gravitated to 3.6 ms, which would mean the panels are delayed 7.5-3.6 or 3.9 ms relative to the subs (maybe some different numbers would make this easier). I had previously been using a 3.3 ms delay as handed down from endless re-thinkings mainly.
I went ahead and put the panels at 10 ms delay. That puts the super tweeters at 9.46 ms (only 0.02 adjustments available) and the subs at 6.10 ms for the same relative delays as above, and gives me full adjustability down to panel delay of up to 10 ms or down to relative negatives just by turning the sub delay knob which turns out to have a very interesting subjective effect.
Then I discovered that running the Tact test at precisely 6.25ms (relative to 10.00 ms in the panels) show the right channel going down (the sub leading) over 100's of MS, and the left channel going up (the sub lagging). This has to be the perfect in-between compromise. At 6.00 ms and 6.5 ms both channels go either up (one more than the other) or down. So I've settled on 6.25 as my best guestimate (equal to 3.75 ms difference from the panels which are delayed by 10 ms for convenience now).
 |
| Tact Impulse Response at 6.25 ms dub delay |
 |
| At 6.50 ms both channels "rising" |
 |
| At 6.0 ms both channels "falling" |
When I play 96kHz, I merely dial up the mid way delay from 10.00 (calibrated at Tact's 48 khz) to 10.38. To play 44.1kHz I should dial down the mid way delay to 9.94 (if I care). To change the bass alignment I can just adjust the bass.
Actually I'm finding that 6.50 msec delay, relative to 10.38 (for 96kHz sampling).or whatever I've set the panels to, works better than 6.25. 6.50 subjectively lets the bass be bass, to hang a bit. 6.25, what I called the "compromise" position, is too dry, the reverse angle taken by the two sides cancel or something and it sounds dry.
Each time I do these alignments via Tact some trick like this arises and gives me an "angle" to make an objective/subjective judgement. Who knows I could be off by more than 1 ms.
Saturday, July 15, 2017
The Dacs I use now are Nearly time aligned (within 0.44 msec)
I finally tested delay, using my ancient test file LoudPolarityTest.wav, which outputs a large scale signal (not -20dB, but I can't remember how high, perhaps -5dB) postive only narrow pulses at about 20 Hz. I couldn't get a clear oscilloscope picture of both dacs at the same time using SpeakerPop, which has much lower frequency of "pops".
At 96kHz, the Emotive and Denon DVD-5000 DACs are nearly perfectly aligned. I found the two traces line up more perfectly adding 0.06msec to the Emotiva Stealth DC-1--which apparently has 0.06msec less latency at 96kHz than the Denon (I'm lucky it's not a much larger difference).
But at 44.1kHz, the alignment slips apart a bit, and the Emotiva there leads by about 0.50 msec. So there is a 0.44 msec difference between the two excess latencies, which is really the important factor in the system design (since I can always align the different ways to accomodate the constant latency differences, at least up to about 300 msec...and in fact I must adjust latency anyway because of speaker offsets, though if doing so by guesswork as I mostly do I would need to keep this in mind). But here the constant latency is actually quite trivial at 0.06 msec (less than one inch sound travel) but the variable latency at 0.44 msec (at more than 6 inches) may be worth thinking about, if not always compensating for.
This isn't perfect, but it's fairly trivial. I can't say my time alignments are necessarily even accurate to 0.44 msec, they may not be "accurate" to 1 msec, though I try for 0.02 msec accuracy when I do such things (not frequently) by measurement, and I have believed rightly or not that I can hear balance differences down to 0.02 msec. I can adjust down to 0.01 msec. I haven't done a measured alignment in years and it could be as much as 2msec off, but I'd guess it is still within about 1msec. Nowadays I set the alignment by guesswork mostly.
However, the lenght of the Acoustat being what it is, time alignment at some point is going to differ as much as 8 msec from time alignment at another part on the same speaker, fwiw.
I could just let this slide, adjusting for best response at 96kHz and leaving that to be that. Or, I could split the difference, leaving a barely audible if at all difference of 0.2 msec on each side (equivalent to about 2 inches of air distance) which is even less noticeable, or I could resample everything to 96 kHz, or I could adjust if I am being super critical. Is it more important that the tweeters align with the panels (you'd think so based on the frequency relationships, but not on the large length of the acoustats, which is going to make any precise alighnment moot, and it's barely audible anyway) or the subs? Right now with DC-1's on subs and tweeters, those are always in the same (mis-) alignment, and they are both point sources, which maybe matters.
I had a lot of concern when just looking at 44.1 whether they were actually within 0.5 msec, or it was 50 msec times some number plus 0.5 msec. But since they line up nearly perfectly at 96kHz, it seems likely there's only a small extra at 44.1 as I observe, and the possibilities of lining up perfectly at 96kHz by chance even though there's a huge difference...seems very unlikely. (I spent much time thinking about this though.)
Testing the RD-V1, which I used for a year before getting the Audio GD Dac 19, it has 0.32 msec less latency than the DC-1, amazing for such an ancient device (there's the Apogee clock, which must be an asynchronous receiver system). AND the latency stays constant at both 44.1 kHz and 96 kHz with the DC-1, which suggests they must both have constant latency (and it was the DVD-5000 et al that have variable latency).
Since I live in the old fashioned world where sources rule because of SPDIF and AES, and Dac's aren't the master with slave control over the source, I put the RD-V1 as my digital transport for CD's, figuring the more accurate clock more important to me there. But, look what it could do as a DAC, perfect time alignment at every rate.
Sadly my recently purchased RDC-7 has blown outputs, that might have been even better than RDV-1 because balanced output and dual differential 1704's. But perhaps I was wrong, actually, in believing the RDV-1 does not have dual differential, as I just read somewhere that it does.
The RDV-1 does not beat the DVD-5000 however in RMAA testing. In fact it's considerably behind in S/N and dynamic range, and about exactly the same in THD. So it would not necessarily be a better mid way dac.
So we have this ultimate jewel configuration of dual differential 1704's in 19 year old DAC's, lots of them, it seems, not as esoteric as Wadia or even Levinson. They vary with state of repair or something. And the analog circuitry, which can generally be assumed to be equivalent to 5534's, which are actually pretty good, I think 5534 distortion is typically below 0.0005%, but not as good as LM 4562. Compared to OPA2134...a very common choice...I'm not sure the 5534's are clearly worse, they're about the same level. The OPA 211 are best of all.
Or maybe my poor measurements of my first DVD-9000 are not a matter of state of repair. The second Denon DVD-9000 measures almost exactly the same 95.8dB S/N ratio, dynamic range, frequency response, the same except distortion in the first DVD-9000 at 0.0014% was actually lower, the second DVD-9000 pretty much matches most of my older dual differential 1704 stuff at 0.003%.
The second DVD-9000 also shows the same hashy looking highest octave noise (but below -110dB) that the first DVD-9000 does. I'm beginning to wonder if that's an artifact of the PMI 100 digital filter being used. I love the sound of the DVD-9000 on HDCD's, that was my original reason for having it, but then I decided it was great on other discs too, except my first wouldn't play DVD-Audio because of a problem Denon Service couldn't fix. Well maybe the smoothing digital filters are actually producing a kind of noise. Perhaps it even sounds good while measuring poorly. But this is all speculation now, though the DVD-9000's are the only players to show that noise, and the only ones with real PMI 100's, it may be coincidence.
Here's a picture showing the 9000's and the 5000, and more:
Clearly the DVD-5000 is the winner as an R2R dac in my collection now. The DVD-9000's both show identical weird top octave noise during the slapped together RMAA test. However there's no reason not to use the newest DVD-9000 as an HDCD player despite higher nominal measured THD which may be measurement artifact from loading differences. The two 9000's differ by the second one having a tad more third harmonic, which contributes to only a tiny rise in IM (because, strangely enough, not an even harmonic). Meanwhile the 5000 is far cleaner at the top, you can see the noise threshold all the way to 20k at least with only a few peaks, but it has more 2nd harmonic, still quite low at -106dB, a flaw I would not consider too serious. The 9000's apparently cancel that better, but have more distotion elsewhere. The 5000 does have a bit more power supply ripple though still -120dB.
The RDV-1 does show much lower jitter as a transport, right at my limits of measurement around 170ps. I think that's what you get from simply tracking a SPDIF stream, the limits of SPDIF itself so to speak, though SPDIF receivers can then reduce it to almost zero using PLL or asynchronous interpolators. The clock inside the RDV-1 is what ultimately matters, and that could be considerably better than 170ps but no worse. Meanwhile, the second DVD-900 measures 500 ps playing the same 880 Hz disc (my current test for this, such as it is), clearly worse, not as good as SPDIF itself.
At 96kHz, the Emotive and Denon DVD-5000 DACs are nearly perfectly aligned. I found the two traces line up more perfectly adding 0.06msec to the Emotiva Stealth DC-1--which apparently has 0.06msec less latency at 96kHz than the Denon (I'm lucky it's not a much larger difference).
But at 44.1kHz, the alignment slips apart a bit, and the Emotiva there leads by about 0.50 msec. So there is a 0.44 msec difference between the two excess latencies, which is really the important factor in the system design (since I can always align the different ways to accomodate the constant latency differences, at least up to about 300 msec...and in fact I must adjust latency anyway because of speaker offsets, though if doing so by guesswork as I mostly do I would need to keep this in mind). But here the constant latency is actually quite trivial at 0.06 msec (less than one inch sound travel) but the variable latency at 0.44 msec (at more than 6 inches) may be worth thinking about, if not always compensating for.
 |
| Initial offset at 44.1kHz, about 50 msec |
 |
| Aligned to compensate in DEQ |
This isn't perfect, but it's fairly trivial. I can't say my time alignments are necessarily even accurate to 0.44 msec, they may not be "accurate" to 1 msec, though I try for 0.02 msec accuracy when I do such things (not frequently) by measurement, and I have believed rightly or not that I can hear balance differences down to 0.02 msec. I can adjust down to 0.01 msec. I haven't done a measured alignment in years and it could be as much as 2msec off, but I'd guess it is still within about 1msec. Nowadays I set the alignment by guesswork mostly.
However, the lenght of the Acoustat being what it is, time alignment at some point is going to differ as much as 8 msec from time alignment at another part on the same speaker, fwiw.
I could just let this slide, adjusting for best response at 96kHz and leaving that to be that. Or, I could split the difference, leaving a barely audible if at all difference of 0.2 msec on each side (equivalent to about 2 inches of air distance) which is even less noticeable, or I could resample everything to 96 kHz, or I could adjust if I am being super critical. Is it more important that the tweeters align with the panels (you'd think so based on the frequency relationships, but not on the large length of the acoustats, which is going to make any precise alighnment moot, and it's barely audible anyway) or the subs? Right now with DC-1's on subs and tweeters, those are always in the same (mis-) alignment, and they are both point sources, which maybe matters.
I had a lot of concern when just looking at 44.1 whether they were actually within 0.5 msec, or it was 50 msec times some number plus 0.5 msec. But since they line up nearly perfectly at 96kHz, it seems likely there's only a small extra at 44.1 as I observe, and the possibilities of lining up perfectly at 96kHz by chance even though there's a huge difference...seems very unlikely. (I spent much time thinking about this though.)
Testing the RD-V1, which I used for a year before getting the Audio GD Dac 19, it has 0.32 msec less latency than the DC-1, amazing for such an ancient device (there's the Apogee clock, which must be an asynchronous receiver system). AND the latency stays constant at both 44.1 kHz and 96 kHz with the DC-1, which suggests they must both have constant latency (and it was the DVD-5000 et al that have variable latency).
Since I live in the old fashioned world where sources rule because of SPDIF and AES, and Dac's aren't the master with slave control over the source, I put the RD-V1 as my digital transport for CD's, figuring the more accurate clock more important to me there. But, look what it could do as a DAC, perfect time alignment at every rate.
Sadly my recently purchased RDC-7 has blown outputs, that might have been even better than RDV-1 because balanced output and dual differential 1704's. But perhaps I was wrong, actually, in believing the RDV-1 does not have dual differential, as I just read somewhere that it does.
The RDV-1 does not beat the DVD-5000 however in RMAA testing. In fact it's considerably behind in S/N and dynamic range, and about exactly the same in THD. So it would not necessarily be a better mid way dac.
So we have this ultimate jewel configuration of dual differential 1704's in 19 year old DAC's, lots of them, it seems, not as esoteric as Wadia or even Levinson. They vary with state of repair or something. And the analog circuitry, which can generally be assumed to be equivalent to 5534's, which are actually pretty good, I think 5534 distortion is typically below 0.0005%, but not as good as LM 4562. Compared to OPA2134...a very common choice...I'm not sure the 5534's are clearly worse, they're about the same level. The OPA 211 are best of all.
Or maybe my poor measurements of my first DVD-9000 are not a matter of state of repair. The second Denon DVD-9000 measures almost exactly the same 95.8dB S/N ratio, dynamic range, frequency response, the same except distortion in the first DVD-9000 at 0.0014% was actually lower, the second DVD-9000 pretty much matches most of my older dual differential 1704 stuff at 0.003%.
The second DVD-9000 also shows the same hashy looking highest octave noise (but below -110dB) that the first DVD-9000 does. I'm beginning to wonder if that's an artifact of the PMI 100 digital filter being used. I love the sound of the DVD-9000 on HDCD's, that was my original reason for having it, but then I decided it was great on other discs too, except my first wouldn't play DVD-Audio because of a problem Denon Service couldn't fix. Well maybe the smoothing digital filters are actually producing a kind of noise. Perhaps it even sounds good while measuring poorly. But this is all speculation now, though the DVD-9000's are the only players to show that noise, and the only ones with real PMI 100's, it may be coincidence.
Here's a picture showing the 9000's and the 5000, and more:
Clearly the DVD-5000 is the winner as an R2R dac in my collection now. The DVD-9000's both show identical weird top octave noise during the slapped together RMAA test. However there's no reason not to use the newest DVD-9000 as an HDCD player despite higher nominal measured THD which may be measurement artifact from loading differences. The two 9000's differ by the second one having a tad more third harmonic, which contributes to only a tiny rise in IM (because, strangely enough, not an even harmonic). Meanwhile the 5000 is far cleaner at the top, you can see the noise threshold all the way to 20k at least with only a few peaks, but it has more 2nd harmonic, still quite low at -106dB, a flaw I would not consider too serious. The 9000's apparently cancel that better, but have more distotion elsewhere. The 5000 does have a bit more power supply ripple though still -120dB.
The RDV-1 does show much lower jitter as a transport, right at my limits of measurement around 170ps. I think that's what you get from simply tracking a SPDIF stream, the limits of SPDIF itself so to speak, though SPDIF receivers can then reduce it to almost zero using PLL or asynchronous interpolators. The clock inside the RDV-1 is what ultimately matters, and that could be considerably better than 170ps but no worse. Meanwhile, the second DVD-900 measures 500 ps playing the same 880 Hz disc (my current test for this, such as it is), clearly worse, not as good as SPDIF itself.
Emotiva!
I've been too disparaging of Emotiva at times. The truth is, I own and rely on many Emotiva products. Because they work and are not stratospherically priced like many others.
I'm relying on my tiny Emotive XPS-1 moving coil phono. It blew away my dB systems dB HG1 even with the extra large power supply. (They both use the same excellent chip, btw, the LT 1115, basically the lowest noise audio IC in general use, and has voltage (but not current) noise slightly lower than my generally preferred OPA 211.) The Emotiva has a sweeter sound, is quieter, and has slightly more gain. I think it has a more sophisticated circuit as well as better parts and an on-board voltage regulator. I true miracle of miniaturized audio design, sold at tiny price. Plus the load selections work very well for me, I choose the 470 ohm loading for my Dynavector 17D3.
The S/N spec is 78dB for moving coil (re: 0.5mV). This is 3dB better than a $2199 Manley Chinook (though I've often and still lust after those).
I've come to believe it might even be about as quiet as the legendary Vendetta SCP-2a. There's nothing in the design of the Vendetta that should necessarily give it better noise performance than a LT 1115 based design. Plus, the Vendetta actually trades away noise performance in various ways, such as putting the high frequency RIAA pole in front of everything, where it can't help noise performance. It does that allegedly to improve IM performance, but it's not clear that is useful either, and the IM spec isn't super outstanding in comparison with the uniqueness of the design. Though I have seen the 90dB spec for the Vendetta, I strongly believe serious specmanship was involved in creating that number...by the more usual 0.5mV A weighted (which itself is a lie, but whatever) 20-20k spec, 78dB is actually incredibly quiet. Cranking up the Tact level to 95.8 I still hear no noise from the XPS-1 (I lost my bid on the Vendetta...good thing as it turned out considering all the needed expenses I've had since).
Anyway, I'm also relying on 2 (!) Steath DC-1's. Since finally dumping the Behringer DCX in favor of Behringer DEQ's for each way, I immediately adopted the Stealth for my subs because it had all the essentials, and good specs, and looked well built. I needed AES input and XLR outputs so subs could theoretically run on a separate circuit (as, in fact, they now do) without ground look issues. And the volume control is a big plus too. Often in other brands you have to pay $2000 to get AES and XLR outputs.
Then, when I (following longstanding prejudices perhaps) decided to use the Denon DVD-5000 as my mid-way DAC (in preference to both the Audio GD Dac 19 and the Stealth DC-1) the DC-1 took its place as the tweeter DAC, in which role it has been outstanding, and in fact it may be quieter, but adds to the realism with its super dynamic sound (possibly my imagination too just because it is so quiet and distortion free), perfect for subs and super tweeters.
And then the Emotive Preamp, XSP-1 (gen 2 I believe) which has made all my Bedroom analog stuff including turntable work for the last few years, and is now proving invaluable as a instrumentation amplifier measuring other stuff...because it's noise and distortion is so low it can usefully measure other equipment.
Now I've devolved my Aragon 28k into a glorified selector switch in my main front end (and removing the tape output series resistors and potentiometer loadings was a big improvement)...I used to use that for instrumentation amplification but it doesn't have balanced outputs and therefore I could never exceed 110dB S/N in instrumentation use and not reliably either. Now with balanced outputs of the Emotiva I can. Plus, of course, I can measure true balanced all the way.
I've heard good things about the amps but I've been more interested in more esoteric options for awhile. If I needed more generic amps (which I now have covered by almost 20 yo Parasound units) I'd consider the Emotivas.
So I love Emotiva and I use them and I'm sad to see the Stealth DC-1 be discontinued just as I'm discovering what a fine product it is. I'd certainly have lower distortion using all 3 DC-1's instead of the other things I have and continue doing, now with 2 DC-1's.
Subscribe to:
Posts (Atom)