Interesting question, but not that easy to get two identical amps with the two different biasing options.Just wondering if there is any thought as to performing listening tests comparing two identical amplifiers, except one with a traditional well implemented Vbe multiplier bias spreader, and the other with this sort of adaptive approach?
Not saying there would be any problem with this circuitry of course. Don't know. Just wondering how it might work in practice with real music and real speakers
The ML No.33 and many other amps in the 300 series having this adaptive biasing are very well reviewed, although this doesn't prove anything about the used biasing system.
But while it is quite easy to simulate the No.20.X and No. 33 output stages in open loop, which is what I already did in #8476, I had let LTSpice do THD calculations with resp. 8R, 4R and 2R all at 25V rms for both output stages. at 3Khz.
Both output stages with the same MJE15030/MJE15031 pre-drivers and drivers and with the same 8 pairs of MJ15024/MJ15025 output transistors, both with the same 2.8 amp biasing current to make the differences as small as possible.
In the attachment, Red is for the ML No.20.X with conventional biasing and Blue for the No.33 topology here also with 8 output pairs.
As can be seen is that with conventional biasing, the relatively harmless even harmonics are lower, but the uneven harmonics are at least 20dB higher for all three load options.
Three things can be commented on this,:
1) All same transistors are 100% identical, which is never the case in the real world, so harmonics will be higher as shown.
2) This test was performed on single output stages in open loop. When situated within an amp, feedback will reduce these harmonics depending on the loop gain.
3) The ML No.33 is in fact a bridged amp, so even harmonics will cancel each other. That's what can be seen in the second attachment, where a true No.33 output stage was simulated with it's two bridged amps.
In general, looking at the spectra, when forced to make a choice, I would tend to prefer the adaptive biasing with >20dB lower uneven harmonics.
Hans
Attachments
This is steady-state FFT result, and the model is driving an resistive load?
Reason I ask is because I have been told that in practice some of these adaptive bias schemes, when interacting with real speakers and cables, and using real music the stimulus, may not be as LTI as presumed by the modeling approach.
IOW, reproduction of dynamics, soundstage, details, etc., may be impacted in a way not well predicted by HD modeling.
EDIT: From looking at some reviews though the actual implementation in ML 33 appears to work well.
Reason I ask is because I have been told that in practice some of these adaptive bias schemes, when interacting with real speakers and cables, and using real music the stimulus, may not be as LTI as presumed by the modeling approach.
IOW, reproduction of dynamics, soundstage, details, etc., may be impacted in a way not well predicted by HD modeling.
EDIT: From looking at some reviews though the actual implementation in ML 33 appears to work well.
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Hi Ed,
Absolutely agree, and I guess you were using an ABYecchs tester? Seriously, doing an ABX comparison this can be a challenge...and as been beaten to death in this forum as well as my own experience conducting them at AES shows; it is not the be-all and end-all of sonic comparison testing methods, and has a few flaws in it's methodology relating to sonic memory.
For me personally (not wanting to start that mess up again) listening to the same section of music repeated with each A, B or X choice can be much more revealing than a synchronized source ABX. Following that methodology, as many different sections of music can be compared for s/n, dynamics/ harmonic content, etc...IMHO it takes a long time and a lot of listening to accurately quantify sonic differences between excellent equipment, if the difference is stark then one or both sources have issues.
Howie
Hi Mark!
I have been using both the Auralic Vega and the RME ADI-2 Pro fs, and both are sonically very good, I have yet to hear drastic differences between them. Many high-end studios are using the RME these days so I had to get one so I could be familiar with it when on the job...and I also find it makes a nice front-end for Jan Didden's Autoranger LE.
What DAC are you listening through these days?
Cheers!
Howie
I have been using both the Auralic Vega and the RME ADI-2 Pro fs, and both are sonically very good, I have yet to hear drastic differences between them. Many high-end studios are using the RME these days so I had to get one so I could be familiar with it when on the job...and I also find it makes a nice front-end for Jan Didden's Autoranger LE.
What DAC are you listening through these days?
Cheers!
Howie
A prototype of Andrea Mori's DSD-only discrete resistor dac.
In the picture there is the dac board with all its resistors, a dual shunt regulator board just at the bottom it, a galvanically isolated FIFO buffer board below that, a transformer output stage board at the top that is shown bypassed when the pic was taken, and two SOA 22/24Mhz clocks in aluminum boxes at the lower left. At the lower right part of the pic, an I2SoverUSB board can be seen. Also included after the pic was taken is an FPGA based PCM->DSD256 converter from a diy project in the Digital Line sub-forum. The dac sounds fantastic. Will destroy your dacs. Seriously. Moreover, soon I will be testing some recent mods that are supposed to make it even better.
In the picture there is the dac board with all its resistors, a dual shunt regulator board just at the bottom it, a galvanically isolated FIFO buffer board below that, a transformer output stage board at the top that is shown bypassed when the pic was taken, and two SOA 22/24Mhz clocks in aluminum boxes at the lower left. At the lower right part of the pic, an I2SoverUSB board can be seen. Also included after the pic was taken is an FPGA based PCM->DSD256 converter from a diy project in the Digital Line sub-forum. The dac sounds fantastic. Will destroy your dacs. Seriously. Moreover, soon I will be testing some recent mods that are supposed to make it even better.
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Howie,
We had an Auralic Vega dac here a few years ago that my friend owned. We compared it to some other dacs including Benchmark DAC-3, another dac I was working on, etc. The Vega without question sounded worst. My friend sold it as soon as he could find a buyer.
Regarding the RME ADI-2 Pro version I am familiar with, it was AK4493 based. Although the RME is well implemented that is at least one step down from AK4499EX, which was not as good as the more recent AK4499EXEQ/AK4191 chipset. I have worked with both versions of AK4499, building dacs that visitors described as better than any oversampling dac they ever heard. Didn't even know it was possible, they said. Both AK4499 versions used hardware conversion of PCM->DSD256 using dual AK4137 ASRC chips that can also do DSD conversion. I went to great lengths to optimize power, bypass, layout, clocking, jitter suppression, galvanic isolation, Vref, output stage, etc. Easily sounded better than the original AK4499EQ based Topping D90, which could also beat either of your dacs in terms of perceptual SQ.
After hearing the prototype of Andrea's new DSD-only dac, I decided to abandon further development of AK4499EXEQ. It simply cannot compete.
Yes, I am using extremely low distortion large panel ESL speakers with exceptionally good mono block power amps and other custom electronics and cabling that you don't have access to. That helps to make it easy to hear differences between dacs.
When I said Andrea's dac could destroy yours, I wasn't making stuff up. Maybe your system isn't good enough to resolve the difference between dacs, buy mine certainly is. If I were to be completely honest, I would describe the Vega as distorted junk by modern standards. You simply have no idea what quality of reproduction is possible. You can't or you wouldn't still be using that thing.
Its possible I invited you in the past to visit Auburn sometime. If not, please consider yourself invited now. Of course, you won't believe what I say until you hear for yourself. I understand that. If you do choose take up my offer, maybe it will be of some use to better inform what you do in your studio work. Again, seriously. Look forward to seeing you if you would be wiling to accept
Sincerely,
Mark
P.S. You would of course be very welcome to bring either or both of your dacs. All the better to see for yourself what I'm trying to explain.
We had an Auralic Vega dac here a few years ago that my friend owned. We compared it to some other dacs including Benchmark DAC-3, another dac I was working on, etc. The Vega without question sounded worst. My friend sold it as soon as he could find a buyer.
Regarding the RME ADI-2 Pro version I am familiar with, it was AK4493 based. Although the RME is well implemented that is at least one step down from AK4499EX, which was not as good as the more recent AK4499EXEQ/AK4191 chipset. I have worked with both versions of AK4499, building dacs that visitors described as better than any oversampling dac they ever heard. Didn't even know it was possible, they said. Both AK4499 versions used hardware conversion of PCM->DSD256 using dual AK4137 ASRC chips that can also do DSD conversion. I went to great lengths to optimize power, bypass, layout, clocking, jitter suppression, galvanic isolation, Vref, output stage, etc. Easily sounded better than the original AK4499EQ based Topping D90, which could also beat either of your dacs in terms of perceptual SQ.
After hearing the prototype of Andrea's new DSD-only dac, I decided to abandon further development of AK4499EXEQ. It simply cannot compete.
Yes, I am using extremely low distortion large panel ESL speakers with exceptionally good mono block power amps and other custom electronics and cabling that you don't have access to. That helps to make it easy to hear differences between dacs.
When I said Andrea's dac could destroy yours, I wasn't making stuff up. Maybe your system isn't good enough to resolve the difference between dacs, buy mine certainly is. If I were to be completely honest, I would describe the Vega as distorted junk by modern standards. You simply have no idea what quality of reproduction is possible. You can't or you wouldn't still be using that thing.
Its possible I invited you in the past to visit Auburn sometime. If not, please consider yourself invited now. Of course, you won't believe what I say until you hear for yourself. I understand that. If you do choose take up my offer, maybe it will be of some use to better inform what you do in your studio work. Again, seriously. Look forward to seeing you if you would be wiling to accept
Sincerely,
Mark
P.S. You would of course be very welcome to bring either or both of your dacs. All the better to see for yourself what I'm trying to explain.
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Ummm... Don't know about that. I am very picky about SQ. Soundstage, dynamics, details, etc., can vary greatly between dacs. One of the worst things about Benchmark DAC-3 was the very narrow and forward soundstage. Well between and in front of the speakers, perceptually speaking. Topping D90 by contrast was wider than the speakers. The large panel ESLs are unforgiving. It not just about HD, THD, IMD, SINAD, etc. The big differences often have little to do with those things.
EDIT: The problem with the Vega as I remember it likely had to do a lot with signal-correlated noise that sounds a lot like distortion, but doesn't look like HD on an FFT. Its not correlated enough to show up as distinct spurs, but can sometimes been seen if one knows what to look for (ESS has called it 'non-PSS' noise, but that's another story).
EDIT: The problem with the Vega as I remember it likely had to do a lot with signal-correlated noise that sounds a lot like distortion, but doesn't look like HD on an FFT. Its not correlated enough to show up as distinct spurs, but can sometimes been seen if one knows what to look for (ESS has called it 'non-PSS' noise, but that's another story).
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I did say DAC-3 was the best ever in terms of measurements available at the time (as could be seen at ASR), and in terms of the best I had heard up until that time. Also at that time I was still using NS-10 speakers and a 20-year old original Bryston 4-B. System has since gone though some improvements...
Learned two things from that experience: (1) Don't trust standard measurements so much, there is more to it than is typically measured, and (2) someone only knows the best they haver heard so far. There is no way to tell what is on the recording versus what is an artifact produced by a dac or other device. At least not until better reproduction and or recording equipment comes along to compare with.
Regarding drumming up sales, I don't care about sales. I get nothing from sales and don't ask for anything. A product should stand on its merits, IMHO.
I merely would like people who want better sound, and who are interested to know if something better has come along to know that it has (not to mention that Howie and I had an extended conversation about a dac project he was once thinking about building, thus I thought he might have interest). Anyway IMHO something newsworthy has indeed come along. Nothing more, nothing less. All the other writing is only bring the incredulous folks back down to Earth. There are good reasons behind the brief summaries I try to give. Unfortunately, brief summaries conjure up all kinds mistaken understandings and or mistaken explanations.
Also, patiently awaiting your visit, Bill. Please do try to find a few hours when you are in the area again
Learned two things from that experience: (1) Don't trust standard measurements so much, there is more to it than is typically measured, and (2) someone only knows the best they haver heard so far. There is no way to tell what is on the recording versus what is an artifact produced by a dac or other device. At least not until better reproduction and or recording equipment comes along to compare with.
Regarding drumming up sales, I don't care about sales. I get nothing from sales and don't ask for anything. A product should stand on its merits, IMHO.
I merely would like people who want better sound, and who are interested to know if something better has come along to know that it has (not to mention that Howie and I had an extended conversation about a dac project he was once thinking about building, thus I thought he might have interest). Anyway IMHO something newsworthy has indeed come along. Nothing more, nothing less. All the other writing is only bring the incredulous folks back down to Earth. There are good reasons behind the brief summaries I try to give. Unfortunately, brief summaries conjure up all kinds mistaken understandings and or mistaken explanations.
Also, patiently awaiting your visit, Bill. Please do try to find a few hours when you are in the area again
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As you know I have 2 small children in tow so even getting everyone to california is a major expense. And when I was last over on my own it was for 48 hours. who knows, one day.
I will note when you say
I will note when you say
that Howie has a side gig building studios so he has more experience than most of us not only of a live mic feed but also how munged it gets before it leaves the studio. Opens up interesting questions over what is good enough.
Absolutely. IMHO a lot of it has to do with economics. How many people can afford a top notch record player? You probably have some idea what that costs. What about top notch speakers? WAF? What can a typical studio afford in the way of space for BIG free standing speakers that need a big room to develop the sound? So on and so on.
What's good enough for 90% - 95% of people isn't necessarily all that good compared to what is possible. In the case of digital audio, what I have found so far is that ADCs are typically a lot better than DACs. Why might that be, given that every modern ADC has a DAC inside of it?
Based on some wise words from MarcelvdG, probably our best dac theoretician here in the forum, it may be because the DACs inside ADCs only have to be accurate at the short window in time when the analog comparator is comparing the analog input to the internal dac output. The dac can glitch, ring, take time to settle, etc., doesn't necessarily matter except for that one time window. OTOH reproduction system dacs are integrated over time. The dac output must be accurate and repeatable at all times, at least the area under the curve must be.
Also, part of what may be involved is that opamps driving ADCs may not be exposed to as much EMI/RFI as opamps at the outputs of modern dac chips. Here is what KSTR had to say about opamps in dacs: https://www.diyaudio.com/community/...wn-by-es9039pro-datasheet.397186/post-7302205 In addition there are some threads over at ASR by PMA that look at conducted and radiated EMI/RFI sensitivity of various opamps.
One beauty of Andrea's dac is that it does not require I/V opamps, nor necessarily differential summing MFB filter opamp stages. Not going into more detail at this point, but have heard what is possible so far. Its better. The limits so far do not seem to be so much in the ADCs used in higher quality recordings.
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@Markw4
Since you do not do controlled listening tests or measurements may I suggest another way to provide us with some objective information: record a track (or passage from a track) with an ADC with both DACs you are comparing. Then post both recordings along with the original. If what you claim holds water the difference should be easible verifiable by analyzing the recordings.
And since you often like to cite Sean Olive this is a good reading:
http://seanolive.blogspot.com/2009/04/dishonesty-of-sighted-audio-product.html
Since you do not do controlled listening tests or measurements may I suggest another way to provide us with some objective information: record a track (or passage from a track) with an ADC with both DACs you are comparing. Then post both recordings along with the original. If what you claim holds water the difference should be easible verifiable by analyzing the recordings.
And since you often like to cite Sean Olive this is a good reading:
http://seanolive.blogspot.com/2009/04/dishonesty-of-sighted-audio-product.html
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There is no evidence that cost relates to quality in audio. There can be a negative correlation. Certainly a highly affordable system can be as good as one costing tens of thousands of dollars in all but subjective ways. (Nothing wrong with that of course, just don't claim it's "better" when it's just a preference).
That can and does happen, but its not always the case. A Lamborghini costs more than a Hugo, but objectively they can both do the speed limit. So it can be argued they are objectively the same if all that is measured is top speed, and both are better than the 'limit' of speed.
However, believing they are truly objectively the same would be rather 'naive.'
To quote Dr. Earl Geddes:
"The bottom line here is that we know so little about how humans perceive the sound quality of an audio system, and in particular the loudspeaker, that one should question almost everything that we think we know about measuring it. From what we have found most of what is being done in this regard is naive."
To quote ESS (the dac chip manufacturer):
There is a slide which initiates discussion about audiophiles with the words, "Understanding what audiophiles are hearing."
"The surprising reality is that sigma-delta DACs can be audibly distinguished from a conventional DAC despite measuring very much better than that DAC."
"...an important point: The human ear detects signals well below the noise level of the DAC."
"The ear is exquisitely sensitive to "unusual" noise sources. Your ancestors camped out by a waterfall (white noise) and yet their 'ears pricked up" when they heard a hint of a predator moving in the undergrowth. (The equivalent visual phenomenon is "seeing something out of the corner of your eye). Noise, to a large degree, can be accommodated by the ear and is not troubling, but the tiniest "anomalous" noise is raised to the conscious level."
"Sigma-delta modulators create non-periodic steady state noise (non-PSS) artifacts..."
"Periodic Steady State analysis is common in RF circuits. It means that the system is forced to repeat a pattern of behavior over and over again with a certain time period. Any artifact is presumed to also repeat in this time period."
"Audio measurements such as THD and DNR are done in the Periodic Steady State. Therefore, they will not activate non-PSS noise. You will not find non-PSS noise by looking at THD, DNR, and SNR."
"As the audio signal moves, the noise does not remain the same."
"Non-PSS noise is the biggest issue, but experiments suggest there are more problems. For example: Audiophiles rate as inferior systems that have variable excess phase noise."
"We find that an unconditionally stable loop sounds better in listening tests."
My comment: The measurements that you may view as objective and complete don't show everything that can be practically measured. Then there are the things we don't know how to measure as well if only using visual assessment of electronic instrument outputs, say, for example, if we want to assess soundstage width. However, I can measure the perceptual width and say that by using my speakers as test instruments the soundstage of Benchmark DAC-3 is about 3-4 feet wide between the speakers and forward. Topping D90, closer to 3x that width, wider than the speakers and more behind them (at least a lower frequencies). In other words there is an approximately 150% wider measured difference. IMHO its no less objective than looking at a scope waveform.
I don't. Still amazes me how many people haven't learned the difference between 'preference' and 'discrimination' in perceptual evaluations. We focus primarily on discrimination here. If we also want to listen for preference then we do that separately.
If you might have some interest in becoming acquainted with perceptual science I would be happy to post some links and references.
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