Eliminating jitter "completely" - Benchmark DAC1 approach
I was intrigued by the apparently *excellent* AP One measurement results for the Benchmark Media DAC1 (available at: http://www.benchmarkmedia.com/digita...%20Rev%20A.PDF ).
I got a reply from their designer and I'm including some information from that reply (I don't want to quote him in entirety as I have not yet received permission to do so). The information provided here is however not a trade secret.
Please look at the measurements before you read further (PDF link above). The jitter immunity this DAC provides is in my personal understanding phenomenal. I have never seen such Audio Precision results for any commercial DAC for which I've seen properly done measurements. The other measures specs are excellent as well.
And yes, I know that jitter immunity isn't be-all-end-all in DAC peformance, but it can be one very important factor.
Avoiding AID is another and I'll get to that one as well later on. Having minimal DC drift in the DA stage is another.
I think these three factors: 1) jitter immunity, 2) AID filtering & 3) DC drift minimisation are perhaps the three most important DA-conversion properties that can make or break the sound of a DAC theoretically. I'm now excluding for the sake of this discussion the analog stage that comes right after the DAC (just trying to concentrate on the actual DA stage).
Please note that I haven't built a DAC like so many of you, my comments are highly theoretical (and not based on a wide experience on various DACs) and I'm willing to accept that they are 'wrong' in real life as opposed to theory.
This is exactly the reason I'm soliciting your more knowledgeable comments on these issues.
Ok, the specifics of DAC1.
- It uses AD1853 DAC ICs in a mode that achieves 115 dB stopband attenuation
- It operates AD1853 at constant 97.65625 kHz sample rate, regardless of input input sample rate, thus shifting the transition band higher (with input sample rate of 48 kHz) than with a sampling rate of 48 kHz would do
- At 96 kHz input rate the fixed clock it only gives DAC1 a transition band shift of c. 860 Hz upwards, still enough to eliminate most transition band problems
- DAC1 uses a FIR low pass filter to incoming PCM stream (before AD1853) with a passband at 44.29 kHz and stopband at 53,37 kHz (achieving 125 dB attenuation)
- It claims to eliminate almost all transition band effects (including aliasing intermodulation distortion effects) for signals with input sample rate of 96 kHz or less. For 192 kHz the data is (I assume) downsampled to 96 kHz and processed further from that.
- I think the measurements (if valid/repeatable) prove this is not just mere marketing hyperbole. The measurements are excellent IMHO. I understand however that measurements are not equal to real life perceived performance.
- It also has a variable volume headphone output (just off the DAC) (op-amps?) with an output impedance of near 0 Ohm to minimize distortion to various headphone loads.
This is coming from the designer, paraphrased by me.
Now, my questions are:
1) Are the approaches used the above 'common' within the DIY DAC community here? Esp. the fixed non-standard sample rate and relate FIR filtering at shifted transition band?
2) Has anybody here actually measure his/her DAC with AP One/Two or Miller Audio Research QC units to assess technical performance? Or does anybody have any measurements from any other DAC to compare?
3) Would $799 USD + courier + VAT of 22% be a too high a price for a professional audio DAC like this, which has gotten a warm response from the few Pro Audio mixers / producers I have been able to get a personal comment from? Do you think the DIY community could equal or outperform this TODAY, that is, is there a DIY design that at least in technical performance is better than what DAC1 claims?
Anything else you wanna add or ask?
I'm asking, because I'm in a market for a new DAC, it needs to be faultless even with jittery sources and can't cost an arm and a leg (well, that's all relative as $800 USD is a quite a chunk of money to me personally).
I'd really appreciate comments on this ready-made DAC approach, even though I understand this is a DIY forum.
All comments welcome.
Re: Eliminating jitter "completely" - Benchmark DAC1 approach
I think oversampling is a very commong method in DIY DACs, and any circuit using sample rate conversion will have a fixed sample rate input to the DAC stage. The non-standard sample rate is just an artifact of the clock speed.
Of course almost all DACs use a FIR filter and have a transition band and many use Analog Devices DAC ICs :)
But I was asking the specfics (level of attenuation at transition band, the amount of transition band shifting and the resulting measured performance).
Can you design a DAC that measures on the bench equally well or better than DAC1?
Can an inexpensive DIY DAC achieve the level of jitter immunity that DAC1 does, which according to measurements I've read beats out Apogee, Aphex, Aardvark, Mark Levinson, Anagram, etc... gear.
The measurements are phenomenal, I'm sure nobody can deny that?
So, are you in fact saying that all this is 'old news' and that most of the oversampling digital filtering DIY DACs around here offer the same level or better performance?
Can you state one example? Can you offer proof in actual AP measurement data?
I really want to know, as I can't buy/build all the DACs and compare them for myself.
This is not meant as a flame, but as an honest question.
PS It's very easy to mouth off, but to prove what one claims... let's see :)
I just don't see anything special about this design. They are using off-the-shelf oscillators, standard three-terminal regulators, and so forth. I'd be very surprised if their UltraLock(TM)(R)(C) technology was anything more than a normal asynchronous sample rate converter.
Also to claim that induced jitter has no effect on the output is meaningless. It could mean the DAC has high jitter rejection, or it could mean that the DAC's self-induced jitter is greater than the jitter on the input.
Ok, fair enough. I trust your judgement.
Thank you for saving me a long penny :)
Now I just need to find the DIY dac that does the same for half the price of less.
Re: Eliminating jitter "completely" - Benchmark DAC1 approach
In the mentioned white paper, they say (or claim?) something like that digital filters are jitter-sensitive. I'm quite a bit confused now 'cause till now I thought (or imagined??) they were not. Could anybody knowledgeable on the issue enlighten me?
Below is an excerpt from the white paper (page 19).
Problem #2: Jitter can severely degrade the anti-alias filters in an oversampling converter. This is a little known
but easily measurable effect. Most audio converters operate at high oversampling ratios. This allows the use of
high-performance digital anti-alias filters in place of the relatively poor performing analog anti-alias filters. In
theory, digital anti-alias filters can have extremely sharp cutoff characteristics, and very few negative effects on
the in-band audio signal. Digital anti-alias filters are usually designed to achieve at least 100 dB of stop-band
attenuation. But, digital filters are designed using the mathematical assumption that the time interval between
samples is a constant. Unfortunately, sample clock jitter in an ADC or DAC varies the effective time interval
between samples. This variation alters the performance of these carefully designed filters. Small amounts of
jitter can severely degrade stop-band performance, and can render these filters useless for preventing aliasing.
Right SKK. This claim is absolutely false. It doesn't matter when or how the bits and words arrive inside the digital filter, as long as data integrity is maintained. The only point where jitter matters in the system is the word clock at the DAC itself. Even the bitclock is irrelevant here.
Digital anti-alias filters... Another joke??
or a real thing? I mean "THE DIGITAL anti-alias filter".
Shouldn't an anti-alias filter be placed before an AD converter in the signal chain? Am I missing something??
If what I think is right, how can possibly an anti-alias filter do its
job *digitally* with analog input? I'm curious..
When I discussed jitter with regards to the AD1853 with the board designer from Analog Devices about 5 years ago, we talked about jitter sensitivity from various sources. It is relatively clear to me that for the DAC chip itself, master clock would likely be the deepest root problem, followed by bitclock and L/R clock.
In audio signals, typically the L/R clock is the biggest problem. The S/PDIF interface is terrible in that the clock is embedded and so there will be significant jitter on both the recovered bit clock and L/R clock. Moreover, the jitter of one will be a function of the jitter of the other.
Kiwi has a good point that indicates that the problem seen by you will depend on the device at hand.
|All times are GMT. The time now is 10:27 AM.|
vBulletin Optimisation provided by vB Optimise (Pro) - vBulletin Mods & Addons Copyright © 2016 DragonByte Technologies Ltd.
Copyright ©1999-2016 diyAudio