What's your budget? I'm partial to the Audio Note balanced DACs for the qualities you mention. Any of the 2.1x, 3.1x and 4.1x Balanced models ought to make you a happy camper.
30 responses Add your response
Jolida JD-100 is a shoe-in for this type sonic character. It is warm, reasonably dynamic, and with very good presence (immediacy). The soundstage is slightly larger, slightly diffuse and slightly less 3-dimensional in comparison to a Musical Fidelity Trivista 21 that I just recently tried in my home but, once the disc starts spinning it's strengths far overshadow any weaknesses - the Jolida is very musical and immenently enjoyable. And, it is not very expensive.
Here's a polarity conundrum. I have a First Sound preamp that inverts polarity. As a result, I reverse the system's polarity at the speaker terminals (+>-, ->+). The Scott Nixon Tube DAC inverts polarity as well. How can I use the TubeDAC in a system with other sources? Seems to me, the TubeDAC signal is going to ultimately be out of phase while the other sources will be in phase, or vice-versa.
Great post Audioengr. I would also add that "smoothness" in many DAC's is achieved at the expense of high frequency articulation i.e. controlled roll-off in the treble region. Not only is tonal balance part of what equates to "smoothness", so does transient response. I think that this is what Audioengr was getting at in terms of "dynamics". A component that lacks transient response will demonstrate a "softer" presentation. While the combination of soft treble response and reduced amplitude of peaks may present a "smoother" presentation, it isn't very accurate or nearly as good sounding as what a well designed digital system is capable of. Sean
Sean, back in 2001 you wrote:
"...most digital reproduction devices seemed to rob the life out of many recordings that i used to love when i was still "vinyling". The introduction of a tube DAC into one of my systems took digital worlds closer to analogue. This added far more space, air, ambience, body, soul and "PRAT" to the system. Instead of sounding like a stripped and sterilized "clone" of a musical recording, i HAD music again. Much like vinyl before it, there was emotion coming from the speakers instead of just a bunch of "digital data" being reproduced. I found it startling to say the least."
You always post very thoughtful comments. I'd like to know what turned you away from the tube DAC you wrote such glowing things about in this post from 2001 to your present preference for SS?
Tvad: I found solid state DAC's that offered all of the benefits of the tubed DAC's but with even greater speed, dimensionality and control. As such, i moved onto the next level. Having said that, most tubed gear sounds much better than most budget or reasonably priced mass produced digital SS gear. As such, i would recommend that someone looking to upgrade their lower cost SS digital gear either check into a tube based DAC or look for a used SS DAC that didn't come off of a mass production assembly line. Sean
Since i seem to be getting on so many people's nerves by commenting on specific products, i'm going to stick to making general comments and technical explanations. The less i say, the better off everybody else will be.
As a side note, the original poster might want to add some further info as to price constraints, specific qualifications ( number and type of inputs required, built in volume control, etc... ). The more info that they provide, the more suitable the suggestions. Sean
Sean, I respect your position to lay low with your opinions. Please know, however, many members (myself included) look to members like yourself to help cut through the jungle of information and make buying decisions clearer. While I like to be educated in elements of good design, I prefer to have clear choices from which I can make a final decision without having to digest spec sheets and technical reports (an aspect of this hobby you appear to enjoy). You can be helpful in that regard at some point if you so choose. Your comments are always welcome (if not keenly circumspect at times :>)
sean or audioengr.
Would you comment on why nonoversanpling DAC's suffer from diminished dynamics and softened transients. I thought the "softening" of transients was due to lack of ringing of the brickwall filter which causes much of the "digititis" we complain about. Why would dynamics be affected by a lack of brickwall filter?
Jayarr: When one looks at a signal at a given spot, it is called "sampling" the data. With digital, these "samples" are taken at very specific but limited points along the data path. One is not taking in ALL of the data, but simply "sampling" it and basing their deductions on averages and trends.
Over-sampling looks at that same data many times i.e. it doesn't look at more data. It does this to verify that that the interpretation of the data actually matches the data that was recovered. As such, over-sampling is more about error correction than it is about increased data recovery.
As far as upsampling goes, this differs from over-sampling. My understanding is that a lower ( standard ) sampling rate provides fewer "check points" of each bit of data. With fewer places along the signal path to extract and compare data, you will suffer from reduced linearity. Reduced linearity results in a loss of information. After all, the less we check up on something and keep track of it, the more that we have to assume about what is going on.
Depending on how and where the data / signal is checked, you might lose resolution on the peaks, between peaks and nulls, etc... By increasing the number of samples taken i.e. up-sampling, there is the potential for less divergence from signal integrity AND an increase in resolution / transparency.
Think of a sine wave i.e. the "hill and valley" looking thing. If we checked the signal at the positive peak ("hill"), in the middle as the signal descends from the peak, at the lowest point of the negative peak ("valley") and back in the middle as the signal ascends again, we could discern quite a bit of information. We would know the absolute amplitude of each peak and the duration of the peaks. From there, we could plot or try to replicate what the signal looked like. If we did this using the data that we recovered from our limited "check points" or sample rates, it would look like a rise accumulating into sharp point falling rapidly to another point that was of opposite or inverted polarity with another rise coming up. Obviously, this looks NOTHING like a sine wave, so we have to guess or "fill in the blanks" in order to approximate what is really taking place. Compared to an analogue signal, which tracks the entire waveform, we've lost TONS of resolution.
By increasing the number of "check points" or sampling rate where data is recorded, we now may be able to discern more of the natural shape of the sine wave. By doing so, we increase linearity by losing less data / having fewer "blank spots" to fill in. We can now tell that the peaks are not necessarily blunt points but rounded mounds. We can also tell that the rise and decay are gradual yet linear radiuses and there are no jagged or abrupt changes in amplitude getting to the top or bottom.
While this explains the increase in "resolution" and / or "transparency" that many experience with a good upsampler, it doesn't explain the difference in amplitudes or dynamic ranges. Or does it?
By producing a higher average signal due to recovering more data, the differences between the positive ( loud ) and negative ( quiet ) passages offers more of an aural contrast. On top of this, we are no longer guessing at how fast the signal rises and falls by filling in the blank spots, we have enough data to know just how fast and where the amplitude of the signal should change at. Hence, the increase in apparent dynamics.
As you mentioned, filtration, bandwidth and transient response all come into play here. Most of this has to do with the analogue section of the Digital to Analogue conversion ( DAC ) taking place. It is also where most of the differences in what we hear with "digital" front ends come into play. By increasing bandwidth, which requires greater speed, and reducing the side-effects of filtering, which also improves transient response ( speed ), there is less blurring or smearing of the signal. Less blurring or smearing with increased bandwidth means better resolution and increased transparency.
Having said that, units and / or formats ( like DVD-A and SACD ) that incorporate higher sampling rates are more likely to recover more of the data. In plain English, this just means that such an approach "lost" less than other, lower samping rates. As such, this is a step forward from "standard" redbook playback.
Units and / or formats that take the side effects of filtering into account and try to minimize / remove them from the signal path are also a step forward. Once again, these are evident in both the DVD-A and SACD formats and redbook players that attempt to do such things.
Each of these on their own are good things. Combining a higher sampling rate with improved linearity due to less filtering / better filtration design can really make a very noticeable improvement. Having said that, redbook players that impliment both of these design advantages are still at a disadvantage compared to either DVD-A and SACD. That is because redbook is limited to working within a given format that is of lower quality to begin with. The other formats offer the potential for much better performance due to building upon a more technologically advanced platform. As it is right now, we are only now "FINALLY" getting to the point where we've learned how to optimize the performance of redbook via manipulating the limitations within this format.
Having said that, the main limitation to any of these formats is in the recording techniques and equipment used to record, mix, master and duplicate the music that we listen to as consumers. As limited as redbook is in bandwidth and "potentially noisy" and "mechanically limited" as vinyl is, i've heard some rather astounding presentations of both. Given that we can do so much with these "limited" formats, i don't think that going to higher resolution playback equipment is the answer or the main problem. That main problem is in the recording / mass production area and you and i won't be able to solve any part of that. That is, unless we start voicing our thoughts / opinions to the record companies / engineers in some manner that they are forced to listen.
Bin: I have an EVS Millennium II in one of my systems. This is an up-sampling design with very limited filtering. My Brother is using an EVS Millennium 1B in his system and my Father is using an EVS Millennium 1A in his system. Both the 1A & 1B are 24/96 units with limited filtering. I have performed some modifications to my unit and that of my Father's, but not my Brother's ( as of yet )... : ) Sean
"Would you comment on why nonoversanpling DAC's suffer from diminished dynamics and softened transients."
Sure. Every CDP and DAC that I have examined (a lot of them) that is non-upsampler has either:
1) Filters on the analog stages that roll-off the HF
2) Compromised power supplies that limit dynamics, particularly at HF
Either one of these will eliminate the digital noise and/or the phasiness at HF, but at the expense of HF extension or dynamics or both. With an upsampler, you get both dynamics and HF extension without digital noise or phasiness.
Sean: "With digital, these "samples" are taken at very specific but limited points along the data path. One is not taking in ALL of the data, but simply "sampling" it and basing their deductions on averages and trends."
No, that's not correct. The Shannon sampling theorem says that your digital samples at 44.1khz or any other data rate will give perfect reconstruction of the analog waveform as long as there are no frequencies present above the Nyquist frequency (which is equal to 1/2 the sampling rate). The reason that oversampling is needed with a 44.1khz signal is not because of poor sampling of the waveform shape***, but rather because of the need to avoid sharp (brickwall) filters in the analog domain. With an oversampled signal, it is possible to use a gentle analog filter and do the steep brickwall cutoff at 22.05 khz in the digital domain, where it is easier to do well.
As far as the difference between oversampling and upsampling, they are the same thing but with different implementation. The traditional 'oversampling', which usually means increasing the sample rate by 8x through interpolation, is done in the d/a chip of the player. The quality of digital filters available in dac ic's is usually not high. When 'upsampling' is used, part of the 8x oversampling (typically 2x or 4x) is done in the processor chip before the data is sent to the d/a chip, and the final 4x or 2x will be done in the d/a chip to complete the total 8x oversampling. The difference here is that the quality of the oversampling filter implemented in the processor can be much higher than those used in the d/a chip.
*** Following your train of thought, if the 44.1khz data samples are not adequately capturing fast transients or peaks, it is necessarily because those peaks include frequencies above Nyquist. Such frequencies can only be recovered with higher basic sample rates (dvd-a, sacd), not by the upsampling of the redbook data. (There are some fine points about jitter, noise and data precision, but I doubt that's what you are arguing.)
Flex: My comments were not meant to come across as up-sampling solving the problem of limited bandwidth / limited transient response inherent in redbook so much as helping to reduce some of those drawbacks to a more bearable ( and even enjoyable ) level.
As far as transient response goes, and as i stated in the above post, many of these limitations are in the analogue conversion / section of the DAC's. Anything that we can do to retrieve more info from the digital portion of reproduction can only increase the resolution of the system as a whole, even if there are losses / degradation in the analogue realm. This is not to say that analogue is inferior to digital, not by any means, but that the implimentation of combining these two different formats is still in its' learning stages.
To be quite honest here, i am still learning about digital and if my comments above are incorrect and / or misleading, please correct me and the mis-information that i may have helped to spread. I passed on info that i learned in these and other forums along with various "hi-fi" articles. As such, it could be wrong or loaded with "half-truths" that need clarification.
Other than that, i do understand and agree with your comments that you made reference to in your "PS" section. That's why i made mention of other formats using higher sampling rates being capable of increased levels of resolution. Sean
I have modded a dAck! and even though it is not an upsampler, it is very nice indeed. Very extended, transparent, good imaging and good dynamics. Best I have heard as the price. Not a match for my upsampling reference DAC though. It is twice as expensive. Next I will be modding a Nixon TubeDAC and a Benchmark DAC1.
The only tube output DAC that I have heard that approaches vinyl is the Audio Note DAC4.1X with mods. Superb sound. Maybe the Nixon has this potential as well