Non-oversamplingDigital filter-less DAC Concept
To Confirm the Original 44.1kHz/16bit Format
It is exciting to create something new. Some people just grab a soldering iron, others deliberately start a simulation. Everyone seems to have his/her own approach. In my case, it starts with going back to the basics, research the history, and re-construct the whole picture in my mind. On starting this project I did research as many resources as I could put my hands on.
As the new generation of CD format is appearing on the horizon, I thought the basic concept should be "To Confirm the Original 44.1kHz/16bit Format". A CD in our hands has exactly the same data, every bit to bit, as to the one that left the studio. To recall this dreamy fact, the above theme would be quite appropriate. Any high-bit or high-sampling does not have its raison d'être unless it surpasses this level of accuracy.
About Non-Oversampling
After examining the following two aspects, I came to a conclusion that 'it is quite difficult to carry out oversampling as theoretically under the current technology'.
1) Oversampling and Jitter
There are two axes on digitizing the sound. The time axis and the amplitude axis. In case of CD, they are 44.1kHz and 16bit. In other words, we have to press in the amplitude data into one of the 16bit stage at every 22.7 s. That produces maximum of +0.5 LSB error, and the digital audio starts by accepting this error at the beginning. However, this error only concerns the amplitude axis and no amount of error was admitted on the time axis. Let me suppose that the accuracy of 16bit means how accurately the acoustic energy (time x amplitude) is transmitted by being distributed into each steps of 16bit. Then, by making the amplitude data more accurate, we can distribute the error onto the time axis.
If we distribute 1/2 of the error,
1 ÷ 44.1kHz ÷ 216; ÷ 2 = 173 (ps)
This represents the maximum limit of the acceptable error (maximum limit of the jitter). (diagram 1)
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[diagram1]
acceptable error of 44.1/16bit | 
[diagram2]
acceptable error of 8 x sampling/20bit |
All of the above is based on the basic sampling rate. When in 8 x oversampling and 20bit, that number would be 1.35ps (diagram 2). This is a totally impossible number to achieve for a separate type DAC which has to recover the clock by PLL. This means that under an average jitter environment, the oversampling can not operate theoretically, and lowers the accuracy within the operating field. In short, just by oversampling the original data, 16bit accuracy can not be satisfied anymore.
2) Oversampling and High-Bit
Originally, oversampling was developed to allow the use of an analog filter with gentler characteristics as a post-filter, and not to increase the amount of information. Many people still misunderstand this.
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[diagram3]
principle of FIR type digital filter |
The principle of the most popular FIR type digital filter is to shift the original data and overlay them together, not to create an additional one (diagram 3).When it overlays the data by multiplying the coefficient to the original data, there appears new information below 16bit and to recover this finer information, we need a higher bit rate processing.
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For example, in case of a high-performance digital filter SM5842, this processing is done in 32bit and the filter round them up to 20bit to the output, creating more errors in the re-quantizing process. Recently, this problem was dealt with and a filter was created which can produce 8 x sampling all at once. But even with that, as long as you can't output the internal word length as it is, there's no way you can prevent the errors to occur.
It may sound contrary, but if you take this error into account, 16bit without oversampling is more accurate than 8x-oversampling/20bit.
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[diagram4] image noise continuation |
Then what is going to happen if you eliminate the oversampling process? Theoretically, the image noise will be repeated infinitely to higher frequencies (diagram 4), and a conventional answer would be 'it will sound awful'. Really? This has nothing to do with the "Shannon's theorem", nor do I intend to challenge that. Shannon's theorem considers a sampling theory on transmitting an information. I am talking about the perception of the information. That is, if I must say, "the limitation of our auditory sense is a powerful low-pass-filter and the Shannon's theorem is satisfied at the echelon of human auditory perception." My challenge is rather toward those who listen to the sound through theories and oscilloscopes.
Another way of thinking is that, even if humans can't hear it, the equipment that follows can and will be affected by it.
However, 8x-oversampling/digital-filter can only cut off the frequencies between 22.05kHz and 330kHz. Everything beyond 330kHz is all coming through untouched, meaning the degree of effect is determined by how the said equipment reacts to the ingredients beyond 330kHz. My guess is, if 100kHz signwave comes through, there won't be any problem.
Problems of the Digital Filter
The diagram 5 shows the principle of the most popular FIR type digital-filter. The "T" represents a delay circuit for each sampling interval, "a" is for the coefficient multiplier, and "+" is an adder. After delaying the input data, it multiplies with the coefficient, and this process is repeated n times. This 'n' is called the number of taps. The more taps it has, the higher the performance of the filter is supposed to be. The delay mentioned above is not that of a calculating time, but more like a waiting time until the next data arrives.
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T:adelay circuit
for each sampling interval
a:coefficient multiplier
+:adder
[diagram5]
FIR type digital filter | 
[diagram6]
FIR type digital filter (in case of SM5842) |
It is rather hard to understand this diagram instinctively. It didn't hit home with me, either. But, one day, it occur to me to replace it with the equivalent of the reproducing hardware system. (diagram 6). The delay circuit is replaced with that of the delay of speed of sound, the multipliers with the attenuators, and the adding is synthesized in the space. The number of the speakers corresponds to that of taps. The diagram shows, as an example, the computation of CD data through the high-performance digital-filter SM5842. The accompanied numbers are the actual sizes in the space when replaced with the hard-ware. Since the sampling frequency of CD is 44.1kHz, each delay time for the 1 x sampling is 22.ms per tap. To achieve 8 x sampling, SM5842 repeats 2 x sampling three times, and each step incorporates the taps of, 169 degrees for 2 x, 29 degrees for 4 x, and 17 degrees for 8 x. The accumulated delay of each step becomes, 1.92ms, 0.16ms, and 0.05ms: total of 2.13ms.
Our auditory sense does the frequency analysis at every 2ms interval, and 2.13ms of delay can be caught by our ear.
If the speed of sound is 346m/s, the total length of the row of speakers becomes 737mm. ( In the diagram, the distance between each speaker is presented by the total delay divided by the total number of taps.)
Now, you can imagine what kind of sound will result from such a system. All the notes coming from the speakers before and behind, will mix, intervene with each other, and spread. I would like to express this expansion of the sound over the time axis as a "diffusion of sound coherence". For example, if an attack of a piano note was not clear enough, as if the felt on the hammer became thicker, you might be hearing this "diffusion of sound coherence".
We also need to consider this issue not only on the playback systems, but more totally, including the recording systems.
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[daiagram7]
in case SM5815A
is used in 1/2 decimation |
The diagram 7 indicates the diaphragm 5 replaced with a recording hardware. If you ever felt the digital recording somewhat lacking a core of the sound, please examine this illustration carefully. In a way, one point recording using digital filter is so much nonsense. The time will come in the near future when the performance of a digital filter will be evaluated not only by its cut-off characteristics but also how small a number of taps it has. If the digital filter is a necessary evil, we have to make sure to limit the total delay within 2ms throughout the recording and playback so that it won't be caught by human auditory sense.
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The Sound of Non-oversampling
We can control the "diffusion of sound coherence" only by constructing it with smaller number of taps. From that aspect, Wadia's decoding computer (13 taps) or Luxman's former fluency DAC, DA-07 (3 taps) are considered to be excellent machines. They both received (Wadia still does) outstanding appraisals at the time for their sensual representation of the sound. The sound of non-oversampling DAC is on the extension of these machines, and theoretically, it can exceed those achievements.
The difference between the non-oversampling DAC and the conventional DAC with the digital filter lies whether you attach importance on the accuracy in the time domain or in the frequency domain. In other words, whether you choose the musical performance or the quality of a sound. This trade-off line defines the boundary of the current digital audio format .
A natural, stress-free sound that communicates the musicians' intention directly to you. That is the sound of non-oversampling DAC. The feel of this sound is closer to that of analog reproduction.
Introducing Non-PLL clock
We can still hear the characteristics of each different transport even after lowering the jitter sensitivity to a minimum by non-oversampling. This is an incomplete section of today's digital audio format. The fundamental advantage of being digital, that its quality does not depend on the conveying form, falls completely short here. That is because we have to create the time axis from the incoming data by PLL at the receiving end. This is a flow inevitable to the current DAI format which requires word-synchronization. It is often misunderstood as if the time axis is digital, instead of analog, because it receives discrete value after sampling, but actually, it is completely analog. When the time axis is distorted, the analog wave form is distorted with it.
Then what is going to happen if we read it by its own clock at the receiving side? Unfortunately, the word-synch can not be kept and the waveform will be broken into shreds. I tried to re-clock it with a separate non-PLL-clock after once it's locked with PLL and reproduced (diagram 8). By doing this, the fluctuations within 1 clock of the PLL are completely absorbed and not transferred to the reproduced waveform. But any fluctuations beyond 1 clock, even if it'[s only 1ps, are magnified into that of 1 clock. This happens often, because of the frequency difference between the PLL clock and non-PLL clock. In case of this 50MHz non-PLL clock, the amount is 20ns per every 0.1ms. This is, after all, more than 100 times of the 16bit criterion.
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1. Non-PLL clocl 50MHz
2. PLL clock 2.8224MHz(44.1kHz x 64fs)
3. re-clock pulse
[daiagram8]
Non-PLL re-clock |
How does it sound? Are the notes broken or jumping around and unbearable to listen to? Somehow it does not, and not only that; it generates an extremely realistic sound field. A certain acoustic atmosphere envelops the room making you feel like you are on the same floor with the performers, communicating the tensions and relaxation among musicians to you.
This experience made me wonder if human ears are somewhat insensible to jitter. Whether it has a large or a small amount of jitter is not really an issue. The real problem is the constant fluctuation of the time axis caused by PLL. What is more important is the structure, including the time axis, of jitter, rather than the amount of it.
However, we can still hear the different characteristics of transports. I suspect that we may be hearing an effect of the original jitter detected by the non-PLL clock and entwined around the beat component.
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Measurements
The diagram 9 indicates the frequency characteristics (with the emphasis on and off). It looks a tube amplifier measurements. The roll off at lower frequencies seems caused by narrow frequency range of the analyzer itself, because I get the same response with whatever CD player I'm measuring (the manufacturer's spec sheet claims 20Hz~100kHz). The fall at the higher frequencies are caused by the aperture effect.
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