Excellent post, Amir. I am reminded of the comparison we did at Music Lovers with Cookie Marenco playing the exact same violin track in RBCD and SACD formats and the SACD sounded *smoother*, sort of buttery, more *analog* as compared to the RBCD. Most in attendance preferred the sound of the SACD, but 3 violin players in attendance all agreed the RBCD replicated the sound of the violin more accurately precisely because of the lack of this seemingly added coloration. Qualifications: exact same gear (pre, amp, speakers, player), matched volume levels, multiple comparisons, but not double blind.
Now how to get Puremusic USB out to the AT&T glass inputs of my 2 DCC2 SEs. Arggggghhhhhhh.
Unfortunately, the article is not a review but a marketing paper for DSD. The existence of a chart or two and a few technical terms does not elevate it to proper description of serious issues with the format which have been clearly documented and discussed: http://sjeng.org/ftp/SACD.pdf
The AES paper above is highly technical so people instead go by the simplistic tactics used in the marketing article mentioned here. Let me see if I can bring the concept down to a level that is easy to understand.
A digital system by definition is faulty as a representation of an analog system. It has a limit of resolution which analog in their does not. Let's look at the extreme case of having a 1 bit system. If I have an input music signal that goes from 0 to 1 volt and I represent it with just a 1 or 0, clearly I have huge amount of distortion. Any signal below 0.5 volt would be zero and anything above a 1. If we convert this to analog, we just get a set of squarewave pulses that represent huge amount of distortion.
As we increase the number of bits, then the distortion goes down. A 2 bit system will represent four levels so our output now is less of a square wave but stepped waveform. It still has huge amount of distortion but clearly, we have reduced it some by having more bits.
The above is not yet related to DSD issues. I will get to that in a minute. But the point is that one has to understand that digital cannot be a prefect representation of an analog system if it cannot have a smooth response that is free of distortion such as described.
The solution to above is called "dither." Dither is essentially noise. By adding this noise to our signal, we randomize the distortion product. Anything multiplied by a random number becomes a random number. So signal dependent distortion in the face of dither, becomes plain noise. While we hear noise, it is not nearly as bothersome as distortion.
Now, if we had a ton of noise in our digital systems, that would not be good either. So what we need here is to have enough noise added as to eliminate distortion, but not too much that all we would hear is noise.
A fancier scheme is called noise-shaping. This says that instead of putting dither noise at all frequencies equally, we could use more high at frequencies > 20 Khz, and hence, less at audible frequencies. This is also talked about in the marketing paper by Andreas.
With me so far? If not, think of dither as salt and pepper you put on your food to taste good!
With multi-bit PCM, such as CD's 16 bits, we have bits we can "donate" to dither. Remember, dither is noise so it needs to add its value to the input signal somehow. Depending on the amount of noise we need to add, we could eat up 1, 2 or even 3 bits of the input signal that way.
This brings us to the issue at hand. DSD uses only 1 bit total. It can be shown mathematically (and done in the paper that I referenced) that in a 1-bit system, we cannot add the amount of dither we need to turn the distortion into noise. Doing so causes the system to saturate like an amplifier clipping, and create distortion. The solution then is to add less and doing so means that we have then let some of the distortions in our digitization remain. Therefore, such a system can never be shown to be distortion free.
I will finish by quoting this bit from the AES paper:
"In recent years, we have seen the consumer audio industry perform a
remarkable feat of salesmanship by proclaiming that 1-bit converters
are better than multi-bit converters, and succeeding in marketing 1-
bit products as preferable for the highest-quality performance. The
original primary motivation for pursuing the 1-bit converter
architecture was not superior performance, but rather the fact that it
is cheaper to manufacture, consumes less power, and can operate
well at the voltages used in battery-powered portable equipment.
This has now become secondary, as 1-bit converters are currently
used in consumer audio equipment at all price and quality levels.
The manufacturers of high-quality converters struggled mightily to
produce 1-bit devices that met the performance goals of the industry.
But, they could never eliminate all the undesirable artefacts of such
converters, and after more than a decade of trying, they came to the
realization that they could produce better performance by using
multi-bit converter architectures in their products. The one inherent
advantage of the 1-bit architecture, namely its avoidance of the levelmatching difficulties found in multi-bit converters, turned out not to
be as significant a benefit as one might have thought. If one
examines the current data-sheets of all the major high-quality
converter manufacturers, one finds that they have almost universally
given up on the 1-bit sigma-delta topology in favor of oversampling
converters using more than two levels. Such converter architectures
can avoid the intractabilities of both the 1-bit and the 20+ -bit
designs. They can be properly dithered, and can thus be guaranteed
to be free of low-level, limit-cycle oscillations (“birdies”).
Moreover, they do not suffer from the high-level instability problems
of the higher-order, 1-bit sigma-delta converters. "
All of this said, I am cool with saying subjectively, DSD sounds great or even superior to PCM. When SACD and DVD-A came out, that was my observation. I have no explanation other than maybe like tubes, R2R and LP, we like these distortions. Or maybe it is the implementation that is at work here. Whatever the cause, it is improper to try to explain how technically DSD is superior to PCM where it can be shown readily that it is not. Indeed, the AES paper has the very same graph and simulations used in Andreas marketing material here but goes on to show how that exact system is subject to distortion and oscillations.
True, to an extent. To the average consumer, however, and to almost all audiophiles who have engaged in these comparisons and proclaimed SACD king, including some of our current membership, it should give one pause for thought.That's not an entirely valid comparison unless one is using a "pure" DSD file like Cookie (or Bruce).
Bruce, have you done a conversion from PCM to DSD and found the latter to sound better?
Just confirming, you started with PCM digital files and conversion to DSD made them sound better than playing as PCM?
DSD uses only 1 bit total. It can be shown mathematically (and done in the paper that I referenced) that in a 1-bit system, we cannot add the amount of dither we need to turn the distortion into noise. Doing so causes the system to saturate like an amplifier clipping, and create distortion. The solution then is to add less and doing so means that we have then let some of the distortions in our digitization remain. Therefore, such a system can never be shown to be distortion free.
All of this said, I am cool with saying subjectively, DSD sounds great or even superior to PCM. When SACD and DVD-A came out, that was my observation. I have no explanation other than maybe like tubes, R2R and LP, we like these distortions. Or maybe it is the implementation that is at work here.
Actually it's easier than you think. I'm using Merging's Emotion/Mykerinos SDIF-3 -> Grimm OC1 -> Playback Designs or EMM Labs DAC via AT&T ST-optical
You'll just need the part in blue
It is not. Multi-bit PCM quantization noise can be completely eliminated because we have so much headroom. Indeed, 24-bit audio is "self-dithering" in that there is natural noise in the right order bits so even out of the box, it is distortion free. If one wanted to have lower noise floor still, the same noise shaping used in DSD can be utilized to push the noise to ultrasonics.I read your post with great interest. However, a couple of things jumped at me, and here's the first... The distortion in a PCM scheme can't be the same or of the same magnitude as DSD, can it?
The AES paper has good simulations of this. It shows that when driving by one or two pure tones, there are harmonic distortions that are just 32 db below the signal! Of course, the frequency by then is well into ultrasonics. So how audible those distortions are is subject to debate. More on this below. PCM on the other hand, has no harmonic distortion so is completely superior in this regard. The bats get to enjoy PCM far more than they do DSD .The above leads one to believe that *audible* digitization distortion remains in the DSD stream, but how true is that at the sampling rates it operates? Is there supporting data on this?
Well, per above, PCM does not have distortion. Due to the fact that we can dither multi-bit PCM, we have no distortion at all. So it is a curious case that a system with distortion would be preferred.Could it also then be that the distortions are simply different, audible and annoying in one case but not in the other?
Andreas' article talks about that. But with high sampling rate PCM, we are free of that in the audible band. This is the point I wanted to make earlier. See this graph from the article:For example, the difference between, say, odd-harmonics vs. even-harmonics? Or could it be that quantization distortion isn't at play here, but rather other things like pre-ringing in PCM brickwall filter implementations which may be absent from DSD?
This is the most important point. The AES article touches on it. It says that maybe the reason we like DSD is because the ultrasonic distortion in our speakers and amps, causes them to sound different in normal audio band.My point is that unless we can understand why DSD may have sounded superior to PCM, is it fair to say that "it is improper to try to explain how technically DSD is superior to PCM where it can be shown readily that it is not"???
Again, high sampling rate PCM should not have an issue here. And at any rate, DSD is often filtered also at playback. If that is a digital filter, it will also ring.And finally, when Bruce converted PCM to DSD and found the latter sounded better than the original, could it be that the lack of distortions like pre-ringing during the conversion to DSD and presumably upon final analog conversion caused the better-than-the-original effect?
As I mentioned, the Koch article is marketing material. It makes statements like "it is proven we can hear 100 Khz" impulse or some such thing. What proof? Proof needs to be presented with references and there is none here. And then he moves on to show the above graph. If we care up to 100 Khz, why would we want the system response to look like that? Did the test that prove 100 Khz was audible use such a distorted response?If so, shouldn't we then be looking at PCM vs DSD and comparing them as complete solutions that include whatever filters, noise-shaping, distortion types et al may be necessary, and not just solely based on quantization distortion? The Koch article certainly appears to be taking such a broader view...
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